- sr-users - kamailio.org

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,

2 1

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

1 1

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/

1 0

Implicit branch creation and private variables questions
by George Diamantopoulos 24 Jan '20

24 Jan '20

Hello all, I'm trying to accomplish the following scenario, and some questions have been raised during testing: I'd like to be able to fork serially to a number of downstream destinations in case of failure, but also try several hosts which are available per destination network before failing over to the next one. To that end, I'm using something similar to the following: request_route { ... $vn(some_var) = "some_value"; $vn(some_other_var) = "some_other_value"; ... $avp(provider_order) = "last_provider"; $avp(provider_order) = "first_provider"; ... t_set_fr(120000, 2000); route(PROVIDER_SELECTION); ... } route[PROVIDER_SELECTION] { if ( is_avp_set("$avp(provider_order)") ) { t_on_failure("PROVIDER_SERIAL"); if ( $avp(provider_priority) == "first_provider" )

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Parallel forking and rtpengine handling
by Sebastian Damm 24 Jan '20

24 Jan '20

Hi, our scenario is the following: We have two clients registered to our Kamailio server, one with a TLS capable phone, one via websocket. Now, when a call comes in, the call is forked and is sent out to both clients. rtpengine handling is done in the branch route, so there are two offers, and we use the "via-branch" parameter. Now, when one branch answers the call, what happens to the other branch? I there a way to delete the other branch? How and in which route? Or does Kamailio do this automatically? I'm happy for a hint. Regards, Sebastian -- Sebastian Damm Voice Engineer __________________________________________ sipgate GmbH

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Re: [SR-Users] TLS verification error LetsEncrypt
by Social Boh 24 Jan '20

24 Jan '20

I'm not sure but with let's encrypt you can create only server certificate, not client certificate so you can't require and verify client certificate. Regards --- I'm SoCIaL, MayBe El 24/01/2020 a las 09:01, Bugaian A. Vitalie escribió: > Ok, thanks. > > But my question is still about why verification fails/or what should > be chked to make it work. Not how to disable it. > > Thanks. > > Vitalie. > > On Fri, Jan 24, 2020 at 2:54 PM Social Boh <social(a)bohboh.info > <mailto:social@bohboh.info>> wrote: > > Hello, > > changing: > > [client:default] > #method = TLSv1.2+ > verify_certificate = yes > > require_certificate = yes > > with > > [client:default] > #method = TLSv1.2+ > verify_certificate = no > require_certificate = no > > --- > I'm SoCIaL, MayBe > > El 24/01/2020 a las 08:46, Bugaian A. Vitalie escribió: >> Hello list, >> >> I have tried to setup my tls config tish LetsEncrypt following >> this post: >> >> https://www.fredposner.com/1836/kamailio-tls-and-letsencrypt/ >> >> My tls config looks like this: >> >> >> [server:default] >> method = TLSv1.2+ >> verify_certificate = no >> require_certificate = no >> private_key = /etc/letsencrypt/live/sbc.example.net-0001/privkey.pem >> certificate = >> /etc/letsencrypt/live/sbc.example.net-0001/fullchain.pem >> ca_list = /etc/letsencrypt/live/sbc.example.net-0001/ca_list.pem >> #ca_list = /usr/local/etc/kamailio/tls/cacert.pem >> #crl = /usr/local/etc/kamailio/tls/crl.pem >> server_name = sbc.example.net <http://sbc.example.net> >> server_id = sbc.example.net <http://sbc.example.net> >> >> #ca_list = /usr/local/etc/fullchain.pem >> #ca_list = /usr/local/etc/kamailio/tls/cacert.pem >> #crl = /usr/local/etc/kamailio/tls/crl.pem >> >> >> # --- >> # This is the default client domain profile. >> # Settings in this domain will be used for all outgoing >> # TLS connections that do not match any other >> # client domain in this configuration file. >> # We require that servers present valid certificate. >> # >> [client:default] >> #method = TLSv1.2+ >> verify_certificate = yes >> require_certificate = yes >> >> =================================== >> My ca_list has all certificates from >> cat /etc/ssl/certs/ca-certificates.crt >> >> /etc/letsencrypt/live/sbcc.example.net/ca_list.pem >> <http://sbcc.example.net/ca_list.pem> >> >> I keep getting certificate validation failed see bellow: >> >> an 24 08:39:56 sbc.example.net <http://sbc.example.net> >> /usr/local/sbin/kamailio[6371]: ERROR: tls [tls_util.h:42]: >> tls_err_ret(): TLS write:error:1416F086:SSL >> routines:tls_process_server_certificate:certificate verify failed >> Jan 24 08:39:56 sbc.example.net <http://sbc.example.net> >> /usr/local/sbin/kamailio[6371]: ERROR: <core> >> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: >> error reading - c: 0x7f0474421f68 r: 0x7f0474422028 (-1) >> Jan 24 08:39:56 sbc.example.net <http://sbc.example.net> >> /usr/local/sbin/kamailio[6370]: ERROR: tls [tls_util.h:42]: >> tls_err_ret(): TLS write:error:1416F086:SSL >> routines:tls_process_server_certificate:certificate verify failed >> Jan 24 08:39:56 sbc.example.net <http://sbc.example.net> >> /usr/local/sbin/kamailio[6370]: ERROR: <core> >> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: >> error reading - c: 0x7f0474401cb8 r: 0x7f0474401d78 (-1) >> >> ===================== >> >> What params should I change or where to look for a solution on >> this one? >> >> Thanks. >> >> Vitalie A. Bugaian. >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org> >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >

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TLS verification error LetsEncrypt
by Bugaian A. Vitalie 24 Jan '20

24 Jan '20

Hello list, I have tried to setup my tls config tish LetsEncrypt following this post: https://www.fredposner.com/1836/kamailio-tls-and-letsencrypt/ My tls config looks like this: [server:default] method = TLSv1.2+ verify_certificate = no require_certificate = no private_key = /etc/letsencrypt/live/sbc.example.net-0001/privkey.pem certificate = /etc/letsencrypt/live/sbc.example.net-0001/fullchain.pem ca_list = /etc/letsencrypt/live/sbc.example.net-0001/ca_list.pem #ca_list = /usr/local/etc/kamailio/tls/cacert.pem #crl = /usr/local/etc/kamailio/tls/crl.pem server_name = sbc.example.net server_id = sbc.example.net #ca_list = /usr/local/etc/fullchain.pem #ca_list = /usr/local/etc/kamailio/tls/cacert.pem #crl = /usr/local/etc/kamailio/tls/crl.pem # --- # This is the default client domain profile. # Settings in this domain will be used for all outgoing # TLS connections that do not match any other # client domain in this configuration file. # We require that servers present valid certificate. # [client:default] #method = TLSv1.2+ verify_certificate = yes require_certificate = yes =================================== My ca_list has all certificates from cat /etc/ssl/certs/ca-certificates.crt >> /etc/letsencrypt/live/ sbcc.example.net/ca_list.pem I keep getting certificate validation failed see bellow: an 24 08:39:56 sbc.example.net /usr/local/sbin/kamailio[6371]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS write:error:1416F086:SSL routines:tls_process_server_certificate:certificate verify failed Jan 24 08:39:56 sbc.example.net /usr/local/sbin/kamailio[6371]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f0474421f68 r: 0x7f0474422028 (-1) Jan 24 08:39:56 sbc.example.net /usr/local/sbin/kamailio[6370]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS write:error:1416F086:SSL routines:tls_process_server_certificate:certificate verify failed Jan 24 08:39:56 sbc.example.net /usr/local/sbin/kamailio[6370]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f0474401cb8 r: 0x7f0474401d78 (-1) ===================== What params should I change or where to look for a solution on this one? Thanks. Vitalie A. Bugaian.

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

24 Jan '20

Hello, after the release of Kamailio 5.3.2 it is time to also release another minor version for the branch 5.2. I plan to release Kamailio 5.2.6 next Friday, 24th January 2020, if there are no new reports of relevant issues that make us to reconsider and postpone. Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

1 1

Ideas to reject a call due to RTPEngine failure when handling the 200Ok
by Patrick Wakano 23 Jan '20

23 Jan '20

Hello list, Hope you all doing well! I have a setup of Kamailio + RTPEngine (so no B2BUA), and I've come to a situation which I can't really figure out the best decision..... The case is a late SDP negotiation that the 200Ok with the SDP offer fails to engage the RTPEngine (for example no RTPEngine available). Also, the same problem can happen with a normal SDP negotiation, in case the INVITE goes through normally but then when engaging the SDP answer for the 200Ok, the RTPengine answer request fails for any reason... In these cases, I can't generate a proper SDP for the 200OOk so the call will connect normally, but the media path will be broken... So for such cases, I am thinking it is reasonable to fail the call. However, failing a call during the 200Ok stage is tricky, at this point we can't convert the 200Ok to a 50X. I tried and Kamailio doesn't allow and also RFC wise this is a no no for a SIP proxy.... I could completely remove the SDP body and forward the 200Ok. This will force a failed SDP negotiation and leave it for the endpoints the task to terminate the call. Is that acceptable? Has anyone faced such situation? Any idea is much appreciated! Kind regards, Patrick Wakano

2 2

uac_replace_from and uac_auth fails to authenticate.
by Kjeld Flarup 23 Jan '20

23 Jan '20

I have a setup where I have a fallback to a GSM number I look up the GSM number and provider information in a database and sets the headers. dlg_manage(); $du = "sip:" + $dbr(ra=>[0,0]); $tu = "sip:"+$rU+"@"+$dbr(ra=>[0,0]); $ru = "sip:"+$rU+"@"+$dbr(ra=>[0,0]); uac_replace_from("sip:"+$dbr(ra=>[0,1])+"@EXTERNALIP"); After this the call goes to a failure_route to do uac_auth() Now my problem is that this works with the providers Asterisk server. But if the call is send to the providers Kamailio server, authentication is rejected. Removing uac_replace_from makes the call accepted on the Kamailio server The only possible problem I can see is that the first INVITE without authentication, has correct From header. But the second with the nonce and auth, uses the wrong From header. Thus two different From headers in the same SIP dialog. Unfortunately uac_replace_from is not allowed in failure_route, so I could test if this is the problem. Is the two different From headers a problem, and how could that be fixed. -- -------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

5 6

ims_ipsec_pcscf mandatory for ims_registrar_pcscf ?
by Hamid Hashmi 23 Jan '20

23 Jan '20

According to Module documentation ims_ipsec_pcscf is not mandatory [cid:9c92ff73-4978-4837-b541-a38e15525140] but mod: ims_registrar_pcscf is giving error on ims_ipsec_pcscf. Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: ims_registrar_pcscf [ims_registrar_pcscf_mod.c:253]: mod_init(): Successfully bound to TM module Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: <core> [core/sr_module.c:632]: find_mod_export_record(): found export of <ul_bind_ims_usrloc_pcscf> in module ims_usrloc_pcscf [/opt/lib64/kamailio/modules/ims_usrloc_pcscf.so] Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: ims_registrar_pcscf [ims_registrar_pcscf_mod.c:264]: mod_init(): Successfully bound to PCSCF Usrloc module Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: <core> [core/sr_module.c:638]: find_mod_export_record(): export of <bind_ims_ipsec_pcscf> not found (flags 0) Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: ERROR: ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: mod_init(): can't bind ims_ipsec_pcscf Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: ERROR: <core> [core/sr_module.c:849]: init_mod(): Error while initializing module ims_registrar_pcscf (/opt/lib64/kamailio/modules/ims_registrar_pcscf.so) Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: cdp_avp [cdp_avp_mod.c:225]: cdp_avp_destroy(): Destroying module cdp_avp Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: WARNING: <core> [core/mem/q_malloc.c:487]: qm_free(): WARNING: free(0) called from cdp_avp: cdp_avp_mod.c: cdp_avp_destroy(226) Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: tcpops [tcpops_mod.c:156]: mod_destroy(): TCP keepalive module unloaded. Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: tm [t_funcs.c:84]: tm_shutdown(): start Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: tm [t_funcs.c:87]: tm_shutdown(): emptying hash table Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: tm [t_funcs.c:89]: tm_shutdown(): removing semaphores Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: tm [t_funcs.c:91]: tm_shutdown(): destroying tmcb lists Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: tm [t_funcs.c:94]: tm_shutdown(): done Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: <core> [core/mem/shm.c:288]: shm_destroy_manager(): destroying memory manager: q_malloc Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: <core> [core/mem/q_malloc.c:1185]: qm_shm_lock_destroy(): destroying the shared memory lock Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: <core> [core/mem/pkg.c:97]: pkg_destroy_manager(): destroying memory manager: q_malloc There is no option to use ims_registrar_pcscf without ims_ipsec_pcscf. [cid:ec21fdae-7d7e-437c-a7ea-6345f36a1428] Regards Hamid R. Hashmi

2 3

rtpengine relay ip and port in avp
by Pafel 23 Jan '20

23 Jan '20

Hello, I am using kamailio 5.3, rtproxy module and sipwise rtpengine. Is there any way I can get the rewritten / relay address and port by the rtpproxy/rtpengine. In avp or some other variable? Regards, Pavel Siderov

2 2

Param.valueat not returning special chars
by Duarte Rocha 21 Jan '20

21 Jan '20

Hello, I've run into a problem while using parameters parsing tools from Kamailio. When i request the value of a parameter it discards some characters like "" for example. Here is an example : $var(Parameters) = "kamailio=\"version5\";sip=hey"; xerr("String Parameter = $var(Parameters)"); xerr("Parameter on position 1 = $(var(Parameters){param.name,0}) and has value = $(var(Parameters){param.valueat,0})"); This code returns : ERROR: <script>: String Parameter = kamailio="version5";sip=hey ERROR: <script>: Parameter on position 0 = kamailio and has value = version5 Is there anything i can do to get the original string? I'm running Kamailio 5.2.0. Best Regards, Duarte Rocha

1 0

Printing xavp
by Ovidiu Sas 21 Jan '20

21 Jan '20

What's the best way to print the entire content of an xavp? The xavp_params_implode() doesn't print the entire stack of an xavp, only the top layer: The following code produces "c=d;e=f;" $xavp(x=>e) = "f"; $xavp(x[0]=>c) = "d"; $xavp(x[1]=>e) = "z"; $xavp(x[1]=>c) = "y"; xavp_params_implode("x", "$var(out)"); Is there a way to print everything inside $xavp(x)? Thanks, Ovidiu

3 10

Re: [SR-Users] Get destination IP from forwarded invite
by CUMHUR KARAHAN 21 Jan '20

21 Jan '20

Hi Joel, Thank you for your response. In fact, I want to get Request-line URI's parameters of the outgoing INVITE that is leaving the first Kamailio box, briefly. Unfortunately "du - destination URL" or "sndto(name)" pseudo-variables indicate the next hope's information. And in this case, the next hope's information is not same as the ClientB's information. As I tried to tell before, our setup is like that: ClientA -> Kamailio1 -> Kamailio2 -> ClientB I need the request line URI parameters of the invite message sent by Kamailio1 to Kamailio2. And these parameters contain ClientB's ip and port information such as 10.252.180.51:36945 as you can see below: INVITE sip:905364167881@10.252.180.51:36945;transport=tls SIP/2.0 Record-Route: <sip:gearbox11.turkcell.tgc:5060;r2=on;lr;did=7f5.f572;nat=yes> Record-Route: <sip:testsip05.tvoip.turkcell.com.tr:443;transport=tls;r2=on;lr;did=7f5.f572;nat=yes> Via: SIP/2.0/UDP gearbox11.turkcell.tgc:5060;branch=z9hG4bK67ad.4e8e863f2475ab8d3053af0593548557.1;i=f Via: SIP/2.0/TLS 100.64.13.139:56993;received=212.252.34.165;branch=z9hG4bK.1SYkXD1Me;rport=32980 From: <sip:905368601286@testsip.tvoip.turkcell.com.tr>;tag=WbINy0ypx To: sip:905364167881@testsip.tvoip.turkcell.com.tr CSeq: 21 INVITE Call-ID: 1wF-vjzGIX ... And, your question: no, we don't use the dispatcher mechanism. We just check the DB to find out Kamailio server which the number registered into and send the messages to this server basically. Thanks for all... -----Original Message----- From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of sr-users-request(a)lists.kamailio.org Sent: 17 Ocak 2020 Cuma 14:00 To: sr-users(a)lists.kamailio.org Subject: sr-users Digest, Vol 176, Issue 17 Send sr-users mailing list submissions to sr-users(a)lists.kamailio.org To subscribe or unsubscribe via the World Wide Web, visit https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users or, via email, send a message with subject or body 'help' to sr-users-request(a)lists.kamailio.org You can reach the person managing the list at sr-users-owner(a)lists.kamailio.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Get destination IP from forwarded invite (CUMHUR KARAHAN) 2. Re: Kamalio + RTP Engine + SIP Client behind NAT (Richard Fuchs) 3. Re: Get destination IP from forwarded invite (Joel Serrano) 4. Re: Kamalio + RTP Engine + SIP Client behind NAT (Nuno Miguel Reis) 5. Re: Kamailio/RTC dinner at Fosdem 2020 (Federico Santulli) 6. Re: http_client interface (Daniel-Constantin Mierla) 7. Re: Kamailio stop to process incoming SIP traffic via TCP. (adeykunov) 8. Question about xhttp module kamailio 5.2.x (Jimmy Rodley) 9. Trouble with 302 Redirect (Jim Leahy) 10. Re: Kamalio + RTP Engine + SIP Client behind NAT (Richard Fuchs) 11. Re: http_client interface (Fred Posner) 12. Re: Trouble with 302 Redirect (Alex Balashov) 13. UCaaS Platform (Dario Vargas) 14. Re: [commercial announcement] (Henning Westerholt) ---------------------------------------------------------------------- Message: 1 Date: Thu, 16 Jan 2020 11:45:15 +0000 From: CUMHUR KARAHAN <cumhur.karahan(a)consultant.turkcell.com.tr> To: "sr-users(a)lists.kamailio.org" <sr-users(a)lists.kamailio.org> Subject: [SR-Users] Get destination IP from forwarded invite Message-ID: <319901a90d17413292ad249eb56df2ec(a)consultant.turkcell.com.tr> Content-Type: text/plain; charset="utf-8" Hi all, I'm a newbie for Kamailio and working on a feature needs end-users' destination IP in the first transferred/forwarded invite message. Basically, our system/call flow is like: ClientA -> Kamailio1 -> Kamailio2 -> ClientB Invite from the clientA: tls:212.252.34.165:32980 tls:172.21.193.100:5061 INVITE sip:905364167881@testsip.tvoip.turkcell.com.tr:443 SIP/2.0 Via: SIP/2.0/TLS 100.64.13.139:56993;branch=z9hG4bK.GoiTxA4Ee;rport From: <sip:905368601286@testsip.tvoip.turkcell.com.tr>;tag=WbINy0ypx To: sip:905364167881@testsip.tvoip.turkcell.com.tr CSeq: 20 INVITE ... Needed Ip (Client B's IP 10.252.180.51 for this case) is in the INVITE row of the forwarded message. Invite to the Kamailio2 : udp:172.21.193.100:5060 udp:172.21.193.98:5060 INVITE sip:905364167881@10.252.180.51:36945;transport=tls SIP/2.0 Record-Route: <sip:gearbox11.turkcell.tgc:5060;r2=on;lr;did=7f5.f572;nat=yes> Record-Route: <sip:testsip05.tvoip.turkcell.com.tr:443;transport=tls;r2=on;lr;did=7f5.f572;nat=yes> Via: SIP/2.0/UDP gearbox11.turkcell.tgc:5060;branch=z9hG4bK67ad.4e8e863f2475ab8d3053af0593548557.1;i=f Via: SIP/2.0/TLS 100.64.13.139:56993;received=212.252.34.165;branch=z9hG4bK.1SYkXD1Me;rport=32980 From: <sip:905368601286@testsip.tvoip.turkcell.com.tr>;tag=WbINy0ypx To: sip:905364167881@testsip.tvoip.turkcell.com.tr CSeq: 21 INVITE ... When the first invite message receives, the dialog variables or sel(msg...) parsed values belong to this first received message, as far as I understand during the tests. However I need the forwarded invite message's variables. Even after t_relay in RELAY route, it looks the message I have is the first one. So, I cannot reach this later IP in the forwarded invite message. As far as I understand, bear with me here please, this IP manipulation executes in t_relay function and in the C side of Kamailio. So, I cannot find a way to do that in the our native scripts. Is there any way to get it without diving C code's abyss? Do you have any idea about it? I would appreciate it, if I can get any help or any clue. Thanks for all... S. Cumhur Karahan [http://s.turkcell.com.tr/SiteAssets/Genel/mail-imza.jpg] <https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=http%3a%2…> Bu elektronik posta ve onunla iletilen butun dosyalar sadece gondericisi tarafindan almasi amaclanan yetkili gercek ya da tuzel kisinin kullanimi icindir. Eger soz konusu yetkili alici degilseniz bu elektronik postanin icerigini aciklamaniz, kopyalamaniz, yonlendirmeniz ve kullanmaniz kesinlikle yasaktir ve bu elektronik postayi derhal silmeniz gerekmektedir. TURKCELL bu mesajin icerdigi bilgilerin do?rulu?u veya eksiksiz oldugu konusunda herhangi bir garanti vermemektedir. Bu nedenle bu bilgilerin ne sekilde olursa olsun iceriginden, iletilmesinden, alinmasindan ve saklanmasindan sorumlu degildir. Bu mesajdaki gorusler yalnizca gonderen kisiye aittir ve TURKCELLin goruslerini yansitmayabilir Bu e-posta bilinen butun bilgisayar viruslerine karsi taranmistir. ________________________________ This e-mail and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you are not the intended recipient you are hereby notified that any dissemination, forwarding, copying or use of any of the information is strictly prohibited, and the e-mail should immediately be deleted. TURKCELL makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the information transmission, reception, storage or use of such in any way whatsoever. The opinions expressed in this message belong to sender alone and may not necessarily reflect the opinions of TURKCELL. This e-mail has been scanned for all known computer viruses. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200116/37a3c32b/…> -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.gif Type: image/gif Size: 92 bytes Desc: image001.gif URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200116/37a3c32b/…> ------------------------------ Message: 2 Date: Thu, 16 Jan 2020 08:23:21 -0500 From: Richard Fuchs <rfuchs(a)sipwise.com> To: sr-users(a)lists.kamailio.org Subject: Re: [SR-Users] Kamalio + RTP Engine + SIP Client behind NAT Message-ID: <4686712f-d865-c6c9-11ac-649966b867b1(a)sipwise.com> Content-Type: text/plain; charset="utf-8"; Format="flowed" On 15/01/2020 13.39, Nuno Miguel Reis wrote: > Hi guys. > > I'm replacing a environment where I was using kamailio + freeswitch by > another where I'm adding rtpengine to the mix. > One of the issues I'm having now is when I have a SIP Client behind NAT: > > When I send the INVITE from the SIP Client, the SDP is using the > private LAN IP address + the advertised media port, let's admit it's > 10.10.10.10:5000 <http://10.10.10.10:5000> when the call establishes > with a public server running kamailio + rtpengine, the actual RTP > arrives from the home router public IP on a natted port, let's admit > it's 100.100.100.100:65100 > <https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=htt > p%3a%2f%2f100.100.100.100%3a65100&umid=39581E1E-9C53-E505-96A6-E057F22 > BD18D&auth=cbd2d006abdb2241308511f72807fe0bc6594b4a-193c6dfa2da7d1ac0c > 93d1a002e73781a0a22d9e>, event though, RTPENGINE assumes that RTP > stream is coming from > 100.100.100.100 > <https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=htt > p%3a%2f%2f100.100.100.100&umid=39581E1E-9C53-E505-96A6-E057F22BD18D&au > th=cbd2d006abdb2241308511f72807fe0bc6594b4a-9e365df6d282d7bf1e79e91e70 > 73c5a94aed6d82>:5000.I'm using the '--sip-source' with RTPENGINE to > make it use the received ip address instead of the private IP coming in the SDP but I'm not finding anything to make RTPENGINE adapt an start sending the RTP FLOW to the port where he starts receiving the RTP flow from, discarding the SDP media port information. Rtpengine does this automatically, unless the `asymmetric` flag is used. You should see a log message about a `confirmed peer address` as soon as the first RTP is received on a port. Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200116/483ec5d0/…> ------------------------------ Message: 3 Date: Thu, 16 Jan 2020 07:23:28 -0800 From: Joel Serrano <joel(a)textplus.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] Get destination IP from forwarded invite Message-ID: <CAMtXxQmmRb-ieA9T+0ofxAxiOd23w_0nHos=nZOy9-JNRahnfA(a)mail.gmail.com> Content-Type: text/plain; charset="utf-8" Hello, If I understand correctly your email, you want the IP/Host of the destination of the outgoing INVITE that is leaving a Kamailio box? First of all, how are you forwarding the request? Are you using dispatcher module? Here are a couple of options: 1- https://www.kamailio.org/wiki/cookbooks/5.3.x/pseudovariables#du_-_destinat… (although this depends on your config, it might not be set) 2- https://www.kamailio.org/wiki/cookbooks/5.3.x/core#onsend_route + https://www.kamailio.org/wiki/cookbooks/5.3.x/pseudovariables#sndto_name If you can describe a little more how you are doing things we might be able to give you more information. Joel. On Thu, Jan 16, 2020 at 3:53 AM CUMHUR KARAHAN < cumhur.karahan(a)consultant.turkcell.com.tr> wrote: > Hi all, > > > > I’m a newbie for Kamailio and working on a feature needs end-users’ > destination IP in the first transferred/forwarded invite message. > > > > Basically, our system/call flow is like: > > ClientA -> Kamailio1 -> Kamailio2 -> ClientB > > > > Invite from the clientA: > > > > *tls:212.252.34.165:32980 > <https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=htt > p%3a%2f%2f212.252.34.165%3a32980&umid=39581E1E-9C53-E505-96A6-E057F22B > D18D&auth=cbd2d006abdb2241308511f72807fe0bc6594b4a-d4b503e38b32fd4dee8 > 59434f30f8c5f3e832496>* > > *tls:172.21.193.100:5061 <http://172.21.193.100:5061>* > > *INVITE sip:905364167881@testsip.tvoip.turkcell.com.tr:443 > <http://sip:905364167881@testsip.tvoip.turkcell.com.tr:443> SIP/2.0* > > *Via: SIP/2.0/TLS 100.64.13.139:56993;branch=z9hG4bK.GoiTxA4Ee;rport* > > *From: <sip:905368601286@testsip.tvoip.turkcell.com.tr > <sip%3A905368601286(a)testsip.tvoip.turkcell.com.tr>>;tag=WbINy0ypx* > > *To: sip:905364167881@testsip.tvoip.turkcell.com.tr > <sip%3A905364167881(a)testsip.tvoip.turkcell.com.tr>* > > *CSeq: 20 INVITE* > > *…* > > > > Needed Ip (Client B’s IP 10.252.180.51 for this case) is in the INVITE > row of the forwarded message. > > > > Invite to the Kamailio2 : > > > > *udp:172.21.193.100:5060 <http://172.21.193.100:5060>* > > *udp:172.21.193.98:5060 <http://172.21.193.98:5060>* > > *INVITE sip:905364167881@**10.252.180.51**:36945;transport=tls > SIP/2.0* > > *Record-Route: > <sip:gearbox11.turkcell.tgc:5060;r2=on;lr;did=7f5.f572;nat=yes>* > > *Record-Route: > <sip:testsip05.tvoip.turkcell.com.tr:443;transport=tls;r2=on;lr;did=7f > 5.f572;nat=yes>* > > *Via: SIP/2.0/UDP > gearbox11.turkcell.tgc:5060;branch=z9hG4bK67ad.4e8e863f2475ab8d3053af0 > 593548557.1;i=f* > > *Via: SIP/2.0/TLS > 100.64.13.139:56993;received=212.252.34.165;branch=z9hG4bK.1SYkXD1Me;r > port=32980* > > *From: <sip:905368601286@testsip.tvoip.turkcell.com.tr > <sip%3A905368601286(a)testsip.tvoip.turkcell.com.tr>>;tag=WbINy0ypx* > > *To: sip:905364167881@testsip.tvoip.turkcell.com.tr > <sip%3A905364167881(a)testsip.tvoip.turkcell.com.tr>* > > *CSeq: 21 INVITE* > > *…* > > > > When the first invite message receives, the dialog variables or > sel(msg…) parsed values belong to this first received message, as far > as I understand during the tests. However I need the forwarded invite message’s variables. > Even after t_relay in RELAY route, it looks the message I have is the > first one. So, I cannot reach this later IP in the forwarded invite > message. As far as I understand, bear with me here please, this IP > manipulation executes in t_relay function and in the C side of > Kamailio. So, I cannot find a way to do that in the our native scripts. > > > > Is there any way to get it without diving C code’s abyss? Do you have > any idea about it? I would appreciate it, if I can get any help or any clue. > > > > Thanks for all... > > > > S. Cumhur Karahan > > > > > <https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=htt > p%3a%2f%2fturkcell.li%2fncfYh&umid=39581E1E-9C53-E505-96A6-E057F22BD18 > D&auth=cbd2d006abdb2241308511f72807fe0bc6594b4a-bbcefdc4443d64fef54812 > 6bd3c5f932d722a0a4> > > Bu elektronik posta ve onunla iletilen butun dosyalar sadece > gondericisi tarafindan almasi amaclanan yetkili gercek ya da tuzel > kisinin kullanimi icindir. Eger soz konusu yetkili alici degilseniz bu > elektronik postanin icerigini aciklamaniz, kopyalamaniz, > yonlendirmeniz ve kullanmaniz kesinlikle yasaktir ve bu elektronik postayi derhal silmeniz gerekmektedir. > > TURKCELL bu mesajin icerdigi bilgilerin doğruluğu veya eksiksiz oldugu > konusunda herhangi bir garanti vermemektedir. Bu nedenle bu bilgilerin > ne sekilde olursa olsun iceriginden, iletilmesinden, alinmasindan ve > saklanmasindan sorumlu degildir. Bu mesajdaki gorusler yalnizca > gonderen kisiye aittir ve TURKCELLin goruslerini yansitmayabilir > > Bu e-posta bilinen butun bilgisayar viruslerine karsi taranmistir. > ------------------------------ > > This e-mail and any files transmitted with it are confidential and > intended solely for the use of the individual or entity to whom they > are addressed. If you are not the intended recipient you are hereby > notified that any dissemination, forwarding, copying or use of any of > the information is strictly prohibited, and the e-mail should > immediately be deleted. > > TURKCELL makes no warranty as to the accuracy or completeness of any > information contained in this message and hereby excludes any > liability of any kind for the information contained therein or for the > information transmission, reception, storage or use of such in any way > whatsoever. The opinions expressed in this message belong to sender > alone and may not necessarily reflect the opinions of TURKCELL. > > This e-mail has been scanned for all known computer viruses. > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200116/682b8d5d/…> -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.gif Type: image/gif Size: 92 bytes Desc: not available URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200116/682b8d5d/…> ------------------------------ Message: 4 Date: Thu, 16 Jan 2020 17:29:53 +0000 From: Nuno Miguel Reis <nmreis(a)student.dei.uc.pt> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] Kamalio + RTP Engine + SIP Client behind NAT Message-ID: <CAGTw3V7Y8pHD+a8psPL7Tah=HTgrmvVRPhPBKB9VCuoujX_Nnw(a)mail.gmail.com> Content-Type: text/plain; charset="utf-8" Hi again. Thanks for all the help and suggestions. I realized the issue happens if using kernel forwarding only. If I change rtpengine to start at userspace without the kernel module enabled everything works fine as expected. Do you have any hints on why this could be happening with the kernel module? I'm running rtpengine like this: usersapece: $ rtpengine -f -L 7 --interface=100.100.100.100 --listen-ng= 127.0.0.1:2223 --tos=184 --sip-source kernel: $ rtpengine -f -L 7 --table=0 --interface=100.100.100.100 --listen-ng=127.0.0.1:2223 --tos=184 --no-fallback --sip-source when using kernel module, I'm also setting this rules on chain INPUT under iptables: -A INPUT -p udp -m udp --dport 5000:16383 -j RTPENGINE --id 0 -A INPUT -p udp -m udp --dport 16384:32768 -j RTPENGINE --id 0 -A INPUT -p udp -m udp --dport 49152:65535 -j RTPENGINE --id 0 Looking forward to hear from you guys. Regards, -- Nuno Miguel Reis Departamento de Engenharia Informática Faculdade de Ciências e Tecnologia Universidade de Coimbra On Thu, Jan 16, 2020 at 1:23 PM Richard Fuchs <rfuchs(a)sipwise.com> wrote: > On 15/01/2020 13.39, Nuno Miguel Reis wrote: > > Hi guys. > > I'm replacing a environment where I was using kamailio + freeswitch by > another where I'm adding rtpengine to the mix. > One of the issues I'm having now is when I have a SIP Client behind NAT: > > When I send the INVITE from the SIP Client, the SDP is using the > private LAN IP address + the advertised media port, let's admit it's > 10.10.10.10:5000 when the call establishes with a public server > running kamailio + rtpengine, the actual RTP arrives from the home > router public IP on a natted port, let's admit it's > 100.100.100.100:65100, event though, RTPENGINE assumes that RTP stream is coming from 100.100.100.100:5000. > I'm using the '--sip-source' with RTPENGINE to make it use the > received ip address instead of the private IP coming in the SDP but > I'm not finding anything to make RTPENGINE adapt an start sending the > RTP FLOW to the port where he starts receiving the RTP flow from, > discarding the SDP media port information. > > Rtpengine does this automatically, unless the `asymmetric` flag is used. > You should see a log message about a `confirmed peer address` as soon > as the first RTP is received on a port. > > Cheers > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200116/d61a6989/…> ------------------------------ Message: 5 Date: Thu, 16 Jan 2020 20:45:04 +0100 From: Federico Santulli <kam.list(a)nhm.it> To: miconda(a)gmail.com, "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Cc: "business(a)lists.kamailio.org" <business(a)lists.kamailio.org> Subject: Re: [SR-Users] Kamailio/RTC dinner at Fosdem 2020 Message-ID: <54CDD921-48D2-47F0-935E-0B181F037774(a)nhm.it> Content-Type: text/plain; charset=us-ascii Hi Daniel, We would like to join the dinner, 3 people from US. Kind regards. Federico Santulli NHM - S.R.L. Via Raffaello Sanzio, 88 81031 Aversa (CE) Italy > Il giorno 9 gen 2020, alle ore 09:52, Daniel-Constantin Mierla <miconda(a)gmail.com> ha scritto: > > 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 -- > https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=www. > kamailioworld.com&umid=39581E1E-9C53-E505-96A6-E057F22BD18D&auth=cbd2d > 006abdb2241308511f72807fe0bc6594b4a-ac4b2b5016612d36d9bf3bf438b6b436b1 > f1ae8e > > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users ------------------------------ Message: 6 Date: Thu, 16 Jan 2020 22:56:03 +0100 From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org>, Fred Posner <fred(a)palner.com> Subject: Re: [SR-Users] http_client interface Message-ID: <4ab3422e-c3d6-7d85-fb3e-53b09bfa6c1a(a)gmail.com> Content-Type: text/plain; charset=utf-8 The libcurl seems to offer an option for that as well: - https://curl.haxx.se/libcurl/c/CURLOPT_INTERFACE.html So it can be a small patch to the module and support it in the future. If you look for a solution on current stable, maybe one of the embedded scripting languages can help, you can execute inline a small script in Lua/Python/Perl with app_lua_run() & equivalents. In the past I used the http client from lua to do more complex http api queries (e.g., with specific headers, ...). Cheers, Daniel On 13.01.20 17:00, Fred Posner wrote: > I'd never had a reason to before, but on a deployment I need http > messages to go out a specific interface. Curl has an --interface > option, but I don't see how to do the same with http_client. Unless > I'm missing something... > > Anyone ever do something like this? > -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=www.kamai… ------------------------------ Message: 7 Date: Thu, 16 Jan 2020 09:49:40 -0700 (MST) From: adeykunov <adeykunov(a)intermedia.net> To: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Kamailio stop to process incoming SIP traffic via TCP. Message-ID: <1579193380553-0.post(a)n5.nabble.com> Content-Type: text/plain; charset=us-ascii Daniel, Got the same issue on 5.3.1 with openssl1.1, debian9. After 3 working days of tests (about ~30-50 wss clients), suddenly we've got a lot of connections stucked in CLOSE_WAIT state. Kamailio called sig_alarm_abort() when we try to reboot. Thanks, Andrey -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html ------------------------------ Message: 8 Date: Thu, 16 Jan 2020 22:02:40 +0000 (UTC) From: Jimmy Rodley <jimmyrodley(a)yahoo.com> To: "sr-users(a)lists.kamailio.org" <sr-users(a)lists.kamailio.org> Subject: [SR-Users] Question about xhttp module kamailio 5.2.x Message-ID: <443353431.8467378.1579212160245(a)mail.yahoo.com> Content-Type: text/plain; charset="utf-8" Hi, I am on kamailio 5.2.4 and want to use the xhttp module for certain management operations. The request involves some DB queries . The DB is remote and via ODBC and might have some delay.The documentation on the xhttp module states about latency http://kamailio.org/docs/modules/devel/modules/xhttp.html mentioning that the request is handled in an asynchronous manner. However, the JSONRPC-S module documentation says that this module implements the support for asynchronous RPC commands only for HTTP and HTTPS transports. https://www.kamailio.org/docs/modules/devel/modules/jsonrpcs.html#idm104941… I am confused about the process under whose context the xttp request . Can there be a dedicated process to handle certain management requests which do some custom instructions like certain DB operations or initiate certain http requests ? Thanks,Rupesh -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200116/f581f9a0/…> ------------------------------ Message: 9 Date: Thu, 16 Jan 2020 15:54:54 -0500 From: Jim Leahy <rickfromsunnyvale(a)gmail.com> To: sr-users(a)lists.kamailio.org Subject: [SR-Users] Trouble with 302 Redirect Message-ID: <CAH3xi0-AEEj_bwLYd4OzSH96Cz8=qyrdsNWTjWnf5Sm1efrvhg(a)mail.gmail.com> Content-Type: text/plain; charset="utf-8" Hi All, I'm having some issues trying to properly send a '302 Moved Temporarily' reply to the calling UAC (UAC1) when the remote UAC (UAC2) returns a 486 status (Do Not Disturb). The goal is to redirect UAC1 to a voicemail server when UAC2 is in DND. Here are the pertinent parts of my config: route { # drop some things right away route(DROP); # perform sanity check route(SANITYCHECK); # handle registrations if(is_method("REGISTER")) { route(REGISTRATION); exit; } # handle invites if(is_method("INVITE")) { route("INVITE"); exit; } route(RELAY); } route[INVITE] { # add this proxy to the record-route so it stays in the loop record_route(); # if we can find this user in the location DB... if(lookup("location")) { # relay the invite to the new dURI t_on_failure("FOURDIGITFAIL"); t_on_reply("FOURDIGITREPLY"); route(RELAY); # go back to the main route return; } onreply_route[FOURDIGITREPLY] { if(t_check_status("4[0-9]{2}")) { t_reply("302", "Moved Temporarily"); } } I'm having issues with the t_reply in the onreply_route. It successfully sends the reply, but I can't seem to set the Contact header properly. I'm trying to set it to the location of the user's mailbox on the voicemail sever (ie sip:1000@vmpbx.domain.com), but nothing I try works. I've tried: remove_hf("Contact"); insert_hf("Contact: sip:1000@vmpbx.domain.com"); t_reply("302", "Moved Temporarily"); But as I understand it, that just changes the header in the reply from UAC2, not the reply I'm sending to UAC1 Then I tried: append_to_reply("Contact: sip:1000@vmpbx.domain.com" ); t_reply("302", "Moved Temporarily"); But I can't use append_to_reply in an onreply block. How can I change the Contact header in the reply that's sent? Am I just going about this the wrong way? Any help would be appreciated. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200116/66fc8ba2/…> ------------------------------ Message: 10 Date: Thu, 16 Jan 2020 17:10:06 -0500 From: Richard Fuchs <rfuchs(a)sipwise.com> To: sr-users(a)lists.kamailio.org Subject: Re: [SR-Users] Kamalio + RTP Engine + SIP Client behind NAT Message-ID: <0b704a3c-afc4-3ca0-8d8b-38e2d70bcae9(a)sipwise.com> Content-Type: text/plain; charset="utf-8"; Format="flowed" On 16/01/2020 12.29, Nuno Miguel Reis wrote: > Hi again. > > Thanks for all the help and suggestions. I realized the issue happens > if using kernel forwarding only. If I change rtpengine to start at > userspace without the kernel module enabled everything works fine as > expected. > Do you have any hints on why this could be happening with the kernel > module? > > I'm running rtpengine like this: > > usersapece: $ rtpengine -f -L 7 --interface=100.100.100.100 > --listen-ng=127.0.0.1:2223 <http://127.0.0.1:2223> --tos=184 --sip-source > > kernel: $ rtpengine -f -L 7 --table=0 --interface=100.100.100.100 > --listen-ng=127.0.0.1:2223 <http://127.0.0.1:2223> --tos=184 > --no-fallback --sip-source > The kernel module receives its instructions from the userspace daemon, so there's no reason there should be a difference. Can you post 1) logs and 2) the flags you use for your offers/answers? Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200116/70d3b6de/…> ------------------------------ Message: 11 Date: Thu, 16 Jan 2020 17:11:23 -0500 From: Fred Posner <fred(a)palner.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] http_client interface Message-ID: <9808e784-3d56-d100-4c46-b136cc09ef08(a)palner.com> Content-Type: text/plain; charset=utf-8 On 1/16/20 4:56 PM, Daniel-Constantin Mierla wrote: > The libcurl seems to offer an option for that as well: > > - https://curl.haxx.se/libcurl/c/CURLOPT_INTERFACE.html > > So it can be a small patch to the module and support it in the future. > > If you look for a solution on current stable, maybe one of the embedded > scripting languages can help, you can execute inline a small script in > Lua/Python/Perl with app_lua_run() & equivalents. In the past I used the > http client from lua to do more complex http api queries (e.g., with > specific headers, ...). > > Cheers, > Daniel > Thank you Daniel, I implemented a work around (little proxy in go) but would love if the option was added for the future. Fred Posner fred(a)qxork.com https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=https%3a%… Direct/SMS: +1 (336) 439-3733 Need Fred? Call Fred. 336-HEY-FRED Matrix: @fred:matrix.lod.com ------------------------------ Message: 12 Date: Thu, 16 Jan 2020 17:31:43 -0500 From: Alex Balashov <abalashov(a)evaristesys.com> To: sr-users(a)lists.kamailio.org Subject: Re: [SR-Users] Trouble with 302 Redirect Message-ID: <20200116223143.GA857@typhoon> Content-Type: text/plain; charset=us-ascii Try doing this in a failure_route[]; onreply_route[] isn't really meant to transform replies, just to passively observe or drop them. Also, you'll want to use append_to_reply(): https://kamailio.org/docs/modules/5.3.x/modules/textops.html#textops.f.appe… Note that the conditions that give rise to the invocation of a failure_route are broader and somewhat more abstract than any given SIP reply -- e.g. transaction reply timeouts. So, you would want to structure your failure_route like this: failure_route[REDIRECT_HANDLE] { if(t_is_canceled()) exit; if(t_branch_timeout()) { # Handle the timeout case... exit; } # Use $T_rpl to access reply code ($rs) and reply reason if needed # ($rs), per: # # https://www.kamailio.org/wiki/cookbooks/5.3.x/pseudovariables#t_rpl_pv if($T_rpl($rs) == 486) { append_to_reply("Contact: <sip:...>\r\n"); t_reply("302", "Moved Temporarily"); exit; } # Some other case. } -- Alex On Thu, Jan 16, 2020 at 03:54:54PM -0500, Jim Leahy wrote: > Hi All, > I'm having some issues trying to properly send a '302 Moved Temporarily' > reply to the calling UAC (UAC1) when the remote UAC (UAC2) returns a 486 > status (Do Not Disturb). The goal is to redirect UAC1 to a voicemail server > when UAC2 is in DND. > Here are the pertinent parts of my config: > > route { > # drop some things right away > route(DROP); > > # perform sanity check > route(SANITYCHECK); > > # handle registrations > if(is_method("REGISTER")) { > route(REGISTRATION); > exit; > } > > # handle invites > if(is_method("INVITE")) { > route("INVITE"); > exit; > } > > route(RELAY); > } > > route[INVITE] { > # add this proxy to the record-route so it stays in the loop > record_route(); > > # if we can find this user in the location DB... > if(lookup("location")) { > # relay the invite to the new dURI > t_on_failure("FOURDIGITFAIL"); > t_on_reply("FOURDIGITREPLY"); > route(RELAY); > # go back to the main route > return; > } > > onreply_route[FOURDIGITREPLY] { > if(t_check_status("4[0-9]{2}")) { > t_reply("302", "Moved Temporarily"); > } > } > > I'm having issues with the t_reply in the onreply_route. It successfully > sends the reply, but I can't seem to set the Contact header properly. I'm > trying to set it to the location of the user's mailbox on the voicemail > sever (ie sip:1000@vmpbx.domain.com), but nothing I try works. I've tried: > remove_hf("Contact"); > insert_hf("Contact: sip:1000@vmpbx.domain.com"); > t_reply("302", "Moved Temporarily"); > > But as I understand it, that just changes the header in the reply from > UAC2, not the reply I'm sending to UAC1 > > Then I tried: > append_to_reply("Contact: sip:1000@vmpbx.domain.com" ); > t_reply("302", "Moved Temporarily"); > > But I can't use append_to_reply in an onreply block. > How can I change the Contact header in the reply that's sent? Am I just > going about this the wrong way? Any help would be appreciated. Thanks > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=http%3a%2…, https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=http%3a%2… ------------------------------ Message: 13 Date: Fri, 17 Jan 2020 00:58:49 -0500 From: Dario Vargas <dariovargas.gcs(a)gmail.com> To: sr-users(a)lists.kamailio.org Subject: [SR-Users] UCaaS Platform Message-ID: <CAHqy4U_jecJALyvoZM6p6oGB5VyC_kZN=Sj3_vqjEhkO-tmORw(a)mail.gmail.com> Content-Type: text/plain; charset="utf-8" Take the LEAD in Unified Communications as a Service! We can help you get started in the next Telco Disruptive Business. White Label Hosted Cloud Platform as a Service (CPaaS) will enable you to offer all of the below products under your own brand - Hosted Contact Center (CCaaS) Hosted Unified Communications (UCaaS) Hosted PBX (Full Feature Set) SIP Trunk Ai Chat Bots Caller Profile Solutions (Display your clients' corporate identity on cell phones) to improve phone answer ratio. Video Surveillance (VSaaS integrated with AWS) Web Conferencing Virtual FAX Business SMS Software developed in the USA to have all the routing features required to connect with US and Canadian carriers for inbound and outbound (LRN, LNP, TF, SMS, Push, Video etc). Use our existing contracts and pricing for telco services or BYOC (bring your own carrier model). Alternately we can provide you with introductions to the carriers of choice below to get started - Nuwave.com - Number 1 in TF 8XX Inbound Voipinvite.com - 1000s of MPS and CPS for SMS and Voice Broadcast Voxbone.com - World Wide DiDs - Low Monthly minimum Peerlessnetwork.com - Outbound / Inbound Business calling Centurylink.com / Level 3 - USA DiDs and outbound and inbound - minimum volumes required Verizon.com - Outbound and Inbound enterprise business calls - minimum volumes required Dario Vargas info(a)dnl-switch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200117/943422d8/…> ------------------------------ Message: 14 Date: Fri, 17 Jan 2020 07:46:13 +0000 From: Henning Westerholt <hw(a)skalatan.de> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org>, Dario Vargas <dariovargas.gcs(a)gmail.com> Subject: Re: [SR-Users] [commercial announcement] Message-ID: <2dcadf18-240a-1eb4-4a46-89d1d84c6007(a)skalatan.de> Content-Type: text/plain; charset="utf-8" Hello Dario, this type of content is not appropriate for the sr-users lists, please post it here. Please use our dedicated business list for this kind of commercial announcements. https://lists.kamailio.org/cgi-bin/mailman/listinfo/business Regards, Henning Am 17.01.20 um 06:58 schrieb Dario Vargas: commercial announcement _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Henning Westerholt - https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=https%3a%… Kamailio services - https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=https%3a%… Kamailio Merchandising - https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=https%3a%… -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200117/fa884666/…> ------------------------------ Subject: Digest Footer _______________________________________________ sr-users mailing list sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users ------------------------------ End of sr-users Digest, Vol 176, Issue 17 *****************************************

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TLS Profiles : server_name and server_id query
by mahesh b 21 Jan '20

21 Jan '20

Hi All, Am using kamailio 5.1.9 version >From http://www.kamailio.org/docs/modules/5.1.x/modules/tls.html , i understand " The value for server_id can be any string, being used to match TLS client config profile, overriding the match on ip:port and server_name. " Which suggests that the server_name and server_id can be different for a tls profile IN my tls.cfg i have [client:default] verify_certificate = no require_certificate = no [client:10.211.160.172:5071] method = TLSv1.2 verify_certificate = yes require_certificate = yes private_key = /ssl/sip/profile1/server.key certificate = /ssl/sip/profile1/server.crt ca_list = /ssl/sip/profile1/ca.crt cipher_list = RSA verify_depth = 9 server_id = profile1 server_name = btip.176.com And in sar.cfg i am doing, before sending the request out $xavp(tls=>server_name)="btip.176.com"; $xavp(tls[0]=>server_id)="profile1"; But if i see the logs , its picking client default profile for tls handshake, Am i missing anything in tls.cfg or sar.cfg ? if i see the code in tls_lookup_cfg() while (p) { if(srvid && srvid->len>0) { LM_DBG("comparing addr: [%s:%d] [%s:%d] -- id: [%.*s] [%.*s]\n", ip_addr2a(&p->ip), p->port, ip_addr2a(ip), port, p->server_id.len, ZSW(p->server_id.s), srvid->len, ZSW(srvid->s)); if(p->server_id.s && p->server_id.len==srvid->len && strncasecmp(p->server_name.s, srvid->s, srvid->len)==0) { LM_DBG("TLS config found by server id\n"); return p; } .... Why is that there is a strncasecmp between p->server_name.s and srvid->s ?? the strncasecmp should be between p->server_id.s and srvid->s right ? Regards, Mahesh.B

2 1

Re: [SR-Users] [EXT] Re: Kamailio 5.2.2 IMS -enum_query question
by Woscek, Martin W. 21 Jan '20

21 Jan '20

Hi Henning, It turned out that I didn't know how an e.164 lookup is initiated particularly the required format of the calling number. Reading through the source code, the function call: is_e164() showed I was missing the leading + char. I also noticed that I missed the LM_ERR in my log files eg: Jan 10 10:50:11 imskamailio /usr/local/sbin/kamailio[24552]: ERROR: enum [enum.c:840]: enum_query(): R-URI user '<1234567>' is not an E164 number In our case we are looking at standing up a ENUM server that can be queried for specific user characteristics to complete a call, so now we append the '+' when we want to check a caller or callee: enum_pv_query("+$rU", "enumsvc.e164.arpa.", "sip+video") enum_pv_query("+$fU", " enumsvc.e164.arpa", "+sip+video") All is good now. Thanks! Martin From: Henning Westerholt <hw(a)skalatan.de> Sent: Saturday, January 18, 2020 6:47 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>; Woscek, Martin W. <mwoscek(a)mitre.org> Subject: [EXT] Re: [SR-Users] Kamailio 5.2.2 IMS -enum_query question Hello Martin, do you see any errors in the log file during the time that you execute the enum_query? Have you tried to test it with activated debug logging? Cheers, Henning Am 10.01.20 um 17:57 schrieb Woscek, Martin W.: Hi, I am trying to invoke unum lookups from calls to enum_query() ,is_from_user_enum() and enum_pv_query(). However we are not seeing any enum lookups over the wire using wireshark. In the Kamailio.cfg the following are set: loadmodule "enum" modparam("enum", "domain_suffix", "mydnsserver.org.") modparam("enum", "i_enum_suffix", " mydnsserver.org.") sample use in our Kamailio.cfg file: if(!enum_query("mydnsserver.org.")) { xlog("....); #do something } else { Xlog("....") #do something else } I do see NAPTR lookups being done by the other Kamailio IMS processes initially when it starts up so I know it has connectivity to the DNS server, but none for the invoked enum queries. We are running Kamailio 5.2.2. Thanks, Martin _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com Kamailio Merchandising - https://skalatan.de/merchandising/

1 0

Re: [SR-Users] Kamailio service error
by Voip support 21 Jan '20

21 Jan '20

I think you need to capture w tcpdump on kamailio port and then check what packet is causing this. tcmdump -f "port 5060" -w file.pcap Then open file in wireshark and try figure out ( i think not valid sip packet coming on 5060 port). wt., 21 sty 2020 o 11:26 Mike from ApoloCore <mike(a)apolocore.com> napisał(a): > I think you need to capture w tcpdump on kamailio port and then check what > packet is causing this. > > tcmdump -f "port 5060" -w file.pcap > > Then open file in wireshark and try figure out ( i think not valid sip > packet coming on 5060 port). > > wt., 14 sty 2020 o 08:54 Alex Balashov <abalashov(a)evaristesys.com> > napisał(a): > >> It would appear that an incoming SIP message has one or more >> grammatically invalid headers that confuse the parser? >> >> — >> Sent from mobile, with due apologies for brevity and errors. >> >> > On Jan 14, 2020, at 2:37 AM, Ratanraj Singh <ratanraj.singh2(a)gmail.com> >> wrote: >> > >> > >> > Hi All, >> > I am troubling with kamailio service error. Can someone help on this >> issue? >> > >> > >> > <image.png> >> > >> > _______________________________________________ >> > Kamailio (SER) - Users Mailing List >> > sr-users(a)lists.kamailio.org >> > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users(a)lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >

1 0

Possible bug in Presence module !?
by SamyGo 21 Jan '20

21 Jan '20

Hi, I'm experiencing a strange behavior with the Kamailio presence module. I've a Polycom phone which takes the the XML tag "version" very seriously and so as soon it SUBSCRIBEs to a parking lot a NOTIFY is sent from server with "version=1" - When the Parking lot/extension gets busy it receives the second NOTIFY with "version=1" whereas it should've been 2. So it fails to light up the BLF icons. Now, any further notify correctly increments the version tag and Polycom blf works fine until its SUBSCRIBE expires and the first cycle of NOTIFY is exchanged. I can tell its a bug because for each NOTIFY the *event_route[tm:local-request]* prints the attribute $subs(version) and it correctly shows the sequence 1,2,3...onwards, However the sngrep/network trace tells it differently, the version shows up as 1,1,2,3...onwards *version:* kamailio 5.2.3 (x86_64/linux) c36229 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, 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_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: c36229 compiled on 22:20:40 Jan 18 2020 with gcc 4.9.2 *MODULES:*#!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_mwi.so" loadmodule "presence_xml.so" loadmodule "presence_dialoginfo.so" loadmodule "presence_reginfo.so" loadmodule "pua.so" loadmodule "pua_rpc.so" loadmodule "pua_dialoginfo.so" modparam("pua", "db_url", DBURL) modparam("pua", "db_mode", 2) modparam("pua", "update_period", 60) modparam("pua", "dlginfo_increase_version", 0) modparam("pua", "reginfo_increase_version", 0) modparam("pua", "check_remote_contact", 1) modparam("pua", "fetch_rows", 1000) modparam("pua", "outbound_proxy", "sip:MY_HOST_IP:MY_HOST_PORT") modparam("pua_dialoginfo", "include_callid", 1) modparam("pua_dialoginfo", "send_publish_flag", FLT_DLGINFO) modparam("pua_dialoginfo", "caller_confirmed", 1) modparam("pua_dialoginfo", "include_tags", 1) modparam("pua_dialoginfo", "override_lifetime", 124) modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence", "db_update_period", 60) modparam("presence", "db_table_lock_type", 0) modparam("presence", "local_log_level", 3) modparam("presence", "notifier_processes", 10) modparam("presence", "force_delete", 0) modparam("presence", "subs_db_mode", 2) modparam("presence", "db_table_lock_type", 0) modparam("presence", "expires_offset", 60) modparam("presence", "send_fast_notify", 1) modparam("presence", "clean_period", 30) modparam("presence", "publ_cache", 0) modparam("presence", "min_expires_action", PRESENCE_MIN_EXPIRES_ACTION) modparam("presence", "min_expires", PRESENCE_MIN_EXPIRES) modparam("presence", "max_expires", PRESENCE_MAX_EXPIRES) modparam("presence", "sip_uri_match", 1) modparam("presence", "waitn_time", 1) modparam("presence", "db_url", DBURL) modparam("presence", "server_address", "sip:MY_HOST_IP:MY_HOST_PORT") modparam("presence_xml", "force_dummy_presence", 1) modparam("presence_xml", "force_active", 1) modparam("presence_xml", "db_url", DBURL) modparam("presence_dialoginfo", "force_single_dialog", 1) modparam("presence_dialoginfo", "force_dummy_dialog", 1) #!endif Looking to find some answers, Thanks, Sammy

2 1

Kamalio + RTP Engine + SIP Client behind NAT
by Nuno Miguel Reis 20 Jan '20

20 Jan '20

Hi guys. I'm replacing a environment where I was using kamailio + freeswitch by another where I'm adding rtpengine to the mix. One of the issues I'm having now is when I have a SIP Client behind NAT: When I send the INVITE from the SIP Client, the SDP is using the private LAN IP address + the advertised media port, let's admit it's 10.10.10.10:5000 when the call establishes with a public server running kamailio + rtpengine, the actual RTP arrives from the home router public IP on a natted port, let's admit it's 100.100.100.100:65100, event though, RTPENGINE assumes that RTP stream is coming from 100.100.100.100:5000. I'm using the '--sip-source' with RTPENGINE to make it use the received ip address instead of the private IP coming in the SDP but I'm not finding anything to make RTPENGINE adapt an start sending the RTP FLOW to the port where he starts receiving the RTP flow from, discarding the SDP media port information. In my previous setup with freeswitch alone, freeswitch did this transparently. How do you guys solve this issue in RTPENGINE? Regards, -- Nuno Miguel Reis Departamento de Engenharia Informática Faculdade de Ciências e Tecnologia Universidade de Coimbra

5 9

rfc: moving lib srutils to core
by Daniel-Constantin Mierla 20 Jan '20

20 Jan '20

Hello, during the last developers meeting in Dusseldorf we had some discussions about simplifying the building process and internal dependencies/libraries. The src/lib/srutils contains now mainly the uuid and json functions, which are used by very common modules such as registrar/usrloc, tm, dialog, jsonrpcs, dmq*, ... There is a part related to the time recurrence matching, used by drouting, tmrec, but the code is not that significant in size, so I am planning to relocate the lib code to the core and get rid of dependencies to internal libsrutils to many modules. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 9-11, 2020, Berlin - www.asipto.com Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

1 0

Ann: Registration to Kamailio World 2020 is open!
by Daniel-Constantin Mierla 20 Jan '20

20 Jan '20

Hello, a quick note to the community forums to announce that the registration for the next Kamailio World Conference (April 27-29, 2020, in Berlin, Germany) is now open! More details are available at: * https://www.kamailioworld.com/k08/registration/ Call for presentations is still open, there is already a group of very interesting submissions, we aim to publish details about them in the near future. Looking forward to meeting many of you in Berlin! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 9-11, 2020, Berlin - www.asipto.com Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

1 0

Kamailio 5.2.2 IMS -enum_query question
by Woscek, Martin W. 18 Jan '20

18 Jan '20

Hi, I am trying to invoke unum lookups from calls to enum_query() ,is_from_user_enum() and enum_pv_query(). However we are not seeing any enum lookups over the wire using wireshark. In the Kamailio.cfg the following are set: loadmodule "enum" modparam("enum", "domain_suffix", "mydnsserver.org.") modparam("enum", "i_enum_suffix", " mydnsserver.org.") sample use in our Kamailio.cfg file: if(!enum_query("mydnsserver.org.")) { xlog("....); #do something } else { Xlog("....") #do something else } I do see NAPTR lookups being done by the other Kamailio IMS processes initially when it starts up so I know it has connectivity to the DNS server, but none for the invoked enum queries. We are running Kamailio 5.2.2. Thanks, Martin

2 1

Question about xhttp module kamailio 5.2.x
by Jimmy Rodley 18 Jan '20

18 Jan '20

Hi, I am on kamailio 5.2.4 and want to use the xhttp module for certain management operations. The request involves some DB queries . The DB is remote and via ODBC and might have some delay.The documentation on the xhttp module states about latency http://kamailio.org/docs/modules/devel/modules/xhttp.html mentioning that the request is handled in an asynchronous manner. However, the JSONRPC-S module documentation says that this module implements the support for asynchronous RPC commands only for HTTP and HTTPS transports. https://www.kamailio.org/docs/modules/devel/modules/jsonrpcs.html#idm104941… I am confused about the process under whose context the xttp request . Can there be a dedicated process to handle certain management requests which do some custom instructions like certain DB operations or initiate certain http requests ? Thanks,Rupesh

2 1

http_client interface
by Fred Posner 17 Jan '20

17 Jan '20

I'd never had a reason to before, but on a deployment I need http messages to go out a specific interface. Curl has an --interface option, but I don't see how to do the same with http_client. Unless I'm missing something... Anyone ever do something like this? -- Fred Posner fred(a)qxork.com https://qxork.com Direct/SMS: +1 (336) 439-3733 Need Fred? Call Fred. 336-HEY-FRED Matrix: @fred:matrix.lod.com

3 4

UCaaS Platform
by Dario Vargas 17 Jan '20

17 Jan '20

Take the LEAD in Unified Communications as a Service! We can help you get started in the next Telco Disruptive Business. White Label Hosted Cloud Platform as a Service (CPaaS) will enable you to offer all of the below products under your own brand - Hosted Contact Center (CCaaS) Hosted Unified Communications (UCaaS) Hosted PBX (Full Feature Set) SIP Trunk Ai Chat Bots Caller Profile Solutions (Display your clients' corporate identity on cell phones) to improve phone answer ratio. Video Surveillance (VSaaS integrated with AWS) Web Conferencing Virtual FAX Business SMS Software developed in the USA to have all the routing features required to connect with US and Canadian carriers for inbound and outbound (LRN, LNP, TF, SMS, Push, Video etc). Use our existing contracts and pricing for telco services or BYOC (bring your own carrier model). Alternately we can provide you with introductions to the carriers of choice below to get started - Nuwave.com - Number 1 in TF 8XX Inbound Voipinvite.com - 1000s of MPS and CPS for SMS and Voice Broadcast Voxbone.com - World Wide DiDs - Low Monthly minimum Peerlessnetwork.com - Outbound / Inbound Business calling Centurylink.com / Level 3 - USA DiDs and outbound and inbound - minimum volumes required Verizon.com - Outbound and Inbound enterprise business calls - minimum volumes required Dario Vargas info(a)dnl-switch.com

2 1

Kamailio stop to process incoming SIP traffic via TCP.
by Kristijan Vrban 16 Jan '20

16 Jan '20

Hi kamailios, i have a creepy situation with v5.2.1 stable Kamilio. After a day or so, Kamailio stop to process incoming SIP traffic via TCP. The incoming TCP network packages get TCP-ACK from the OS (Debian 9, 4.18.0-15-generic-Linux) but Kamailio does not show any processing for the SIP-Traffic incoming via TCP. No logs, nothing. While traffic via UDP is working just totally fine. When i look via command "netstat -ntp" is see, that the Recv-Q get bigger and bigger. e.g.: Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name tcp 4566 0 172.17.217.12:5060 xxx.xxx.xxx.xxx:57252 ESTABLISHED 31347/kamailio After Kamailio restart, all is working fine again for a day. We have maybe 10-20 devices online via TCP and low call volume (1-2 call per minute). The only settings for tcp we have is "tcp_delayed_ack=no" How to could we debug this situation? Again, no error, no warings in the log. Just nothing. Kristijan

11 46

Get destination IP from forwarded invite
by CUMHUR KARAHAN 16 Jan '20

16 Jan '20

Hi all, I'm a newbie for Kamailio and working on a feature needs end-users' destination IP in the first transferred/forwarded invite message. Basically, our system/call flow is like: ClientA -> Kamailio1 -> Kamailio2 -> ClientB Invite from the clientA: tls:212.252.34.165:32980 tls:172.21.193.100:5061 INVITE sip:905364167881@testsip.tvoip.turkcell.com.tr:443 SIP/2.0 Via: SIP/2.0/TLS 100.64.13.139:56993;branch=z9hG4bK.GoiTxA4Ee;rport From: <sip:905368601286@testsip.tvoip.turkcell.com.tr>;tag=WbINy0ypx To: sip:905364167881@testsip.tvoip.turkcell.com.tr CSeq: 20 INVITE ... Needed Ip (Client B's IP 10.252.180.51 for this case) is in the INVITE row of the forwarded message. Invite to the Kamailio2 : udp:172.21.193.100:5060 udp:172.21.193.98:5060 INVITE sip:905364167881@10.252.180.51:36945;transport=tls SIP/2.0 Record-Route: <sip:gearbox11.turkcell.tgc:5060;r2=on;lr;did=7f5.f572;nat=yes> Record-Route: <sip:testsip05.tvoip.turkcell.com.tr:443;transport=tls;r2=on;lr;did=7f5.f572;nat=yes> Via: SIP/2.0/UDP gearbox11.turkcell.tgc:5060;branch=z9hG4bK67ad.4e8e863f2475ab8d3053af0593548557.1;i=f Via: SIP/2.0/TLS 100.64.13.139:56993;received=212.252.34.165;branch=z9hG4bK.1SYkXD1Me;rport=32980 From: <sip:905368601286@testsip.tvoip.turkcell.com.tr>;tag=WbINy0ypx To: sip:905364167881@testsip.tvoip.turkcell.com.tr CSeq: 21 INVITE ... When the first invite message receives, the dialog variables or sel(msg...) parsed values belong to this first received message, as far as I understand during the tests. However I need the forwarded invite message's variables. Even after t_relay in RELAY route, it looks the message I have is the first one. So, I cannot reach this later IP in the forwarded invite message. As far as I understand, bear with me here please, this IP manipulation executes in t_relay function and in the C side of Kamailio. So, I cannot find a way to do that in the our native scripts. Is there any way to get it without diving C code's abyss? Do you have any idea about it? I would appreciate it, if I can get any help or any clue. Thanks for all... S. Cumhur Karahan [http://s.turkcell.com.tr/SiteAssets/Genel/mail-imza.jpg] <http://turkcell.li/ncfYh> Bu elektronik posta ve onunla iletilen butun dosyalar sadece gondericisi tarafindan almasi amaclanan yetkili gercek ya da tuzel kisinin kullanimi icindir. Eger soz konusu yetkili alici degilseniz bu elektronik postanin icerigini aciklamaniz, kopyalamaniz, yonlendirmeniz ve kullanmaniz kesinlikle yasaktir ve bu elektronik postayi derhal silmeniz gerekmektedir. TURKCELL bu mesajin icerdigi bilgilerin do?rulu?u veya eksiksiz oldugu konusunda herhangi bir garanti vermemektedir. Bu nedenle bu bilgilerin ne sekilde olursa olsun iceriginden, iletilmesinden, alinmasindan ve saklanmasindan sorumlu degildir. Bu mesajdaki gorusler yalnizca gonderen kisiye aittir ve TURKCELLin goruslerini yansitmayabilir Bu e-posta bilinen butun bilgisayar viruslerine karsi taranmistir. ________________________________ This e-mail and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you are not the intended recipient you are hereby notified that any dissemination, forwarding, copying or use of any of the information is strictly prohibited, and the e-mail should immediately be deleted. TURKCELL makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the information transmission, reception, storage or use of such in any way whatsoever. The opinions expressed in this message belong to sender alone and may not necessarily reflect the opinions of TURKCELL. This e-mail has been scanned for all known computer viruses.

2 1

Negative ACK issue
by Michael Broughton 15 Jan '20

15 Jan '20

Hello, Long time Kam/Ser user, first time poster. I'm running into a problem with one of our telco providers when we make a call that ends up being not in service or some other error. In this case our ACK's are not working and the phone line stays open for a period of time until something times out on their end. They claim the issue is that our negative ACK message is dropping one of the Via headers. This is the only case I can find in our setup where Kamailio does this. But it does drop the first Via, which is the first hop in our internal network. I don't understand why this is a problem for them, and I'm still trying to get a reasonable explanation out of them. Technically, I don't see why it would be a problem. This behaviour is not an issue with our other telco providers. Strangely enough, it is also not an issue for this provider when we make the calls over their MPLS network (we are switching to the internet). My question is, can this behaviour be changed in Kamailio somehow? Is there a way for it to keep all the Via headers for negative ACK's? Or, do I just need to poke them harder to fix their issues? My setup: Source -> Kamailio -> Firewall (NAT, SIP ALG) -> Telco Provider I hope I have provided enough information. Thanks! Michael

4 9

Kamailio Slack
by Abdirahman A. Osman 15 Jan '20

15 Jan '20

Hello, Is the Kamailio Slack open for sr-users , if yes where can I find the invite link? Thanks Abdirahman

3 2

Re: [SR-Users] sr-users Digest, Vol 176, Issue 11
by Sam Ware 14 Jan '20

14 Jan '20

Alex, I get the following error ERROR: <core> [core/pvapi.c:1127]: pv_parse_spec2(): wrong char [ ERROR: bad config file (1 errors) CRITICAL: <core> [core/cfg.y:3537]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 462, column 23-40: Can't get from cache: $(rU{s.select,0,#) David, I did try escaping and get the same error. $avp(techprefix) = $(rU{s.select,0,\#}); additionally, I think I initially tried it double quoted. $avp(techprefix) = $(rU{s.select,0,"#"}); Resulted in: ERROR: <core> [core/pvapi.c:1127]: pv_parse_spec2(): wrong char [ ERROR: bad config file (1 errors) CRITICAL: <core> [core/cfg.y:3537]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 462, column 23-42: Can't get from cache: $(rU{s.select,0,"#") Here is the entire block. if ($rU =~ "[\*\#]") { xlog("L_INFO", "[$ci] -> $rU appears to have a tech prefix, processing...\n"); if ($rU =~ "\*") { $avp(techprefix) = $(rU{s.select,0,*}); ## Reset the RURI user to the non-tech prefix'ed value $rU = $(rU{s.select,1,*}); ## Overwrite the message source variable with "sourceip-techprefix" $var(messagesource) = $_s($si-$avp(techprefix)); } else if ($rU =~ "\#") { $avp(techprefix) = $(rU{s.select,0,#)}); ## Reset the RURI user to the non-tech prefix'ed value $rU = $(rU{s.select,1,#}); ## Overwrite the message source variable with "sourceip-techprefix" $var(messagesource) = $_s($si-$avp(techprefix)); } } On Sat, Jan 11, 2020 at 5:00 AM <sr-users-request(a)lists.kamailio.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.kamailio.org > > You can reach the person managing the list at > sr-users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > Today's Topics: > > 1. Re: Kamailio in Docker Compile Error (Sergey Safarov) > 2. Re: Kamailio in Docker Compile Error (Sam Ware) > 3. Kamailio 5.2.2 IMS -enum_query question (Woscek, Martin W.) > 4. String Transformation Question (Sam Ware) > 5. Re: String Transformation Question (Alex Balashov) > 6. Re: String Transformation Question (David Villasmil) > > > > ---------- Forwarded message ---------- > From: Sergey Safarov <s.safarov(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> > Cc: > Bcc: > Date: Fri, 10 Jan 2020 14:38:02 +0300 > Subject: Re: [SR-Users] Kamailio in Docker Compile Error > please try kamailio-ci:5.3.2-alpine > I will fix error about kamailio-ci:5.3.1-alpine. > > On Fri, Jan 10, 2020 at 1:53 PM Asgaroth <00asgaroth00(a)gmail.com> wrote: > >> Hi, >> >> I see that the kamailio-ci:5.3.1-alpine tag exists, but was getting an >> error when trying to pull it 2 days ago, 5.3.0- alpine works fine. >> >> Unfortunately I cannot post the actual docker pull error right now (in >> airport, on my phone) >> >> On Fri 10 Jan 2020, 08:32 Sergey Safarov, <s.safarov(a)gmail.com> wrote: >> >>> yum -y install yum-utils >>> yum-config-manager --add-repo http://rpm.kamailio.org/centos/kamailio.repo >>> yum install kamailio >>> >>> >>> Also you can use this docker image >>> https://hub.docker.com/repository/docker/kamailio/kamailio-ci >>> >>> On Fri, Jan 10, 2020 at 10:58 AM Daniel-Constantin Mierla < >>> miconda(a)gmail.com> wrote: >>> >>>> Hello, >>>> >>>> try to install which and pkg-config tools. >>>> >>>> Cheers, >>>> Daniel >>>> On 10.01.20 02:21, Sam Ware wrote: >>>> >>>> I am attempting to create my own docker image on Centos 7. During the >>>> build, I get the following error message during the "make all" phase. >>>> ---------------------------------------------------------------------- >>>> CC (gcc) [kamailio] core/cfg.tab.o >>>> LD (gcc) [kamailio] kamailio >>>> make[1]: which: Command not found >>>> make[1]: which: Command not found >>>> make[1]: which: Command not found >>>> make[1]: which: Command not found >>>> make[1]: which: Command not found >>>> make[1]: which: Command not found >>>> CC (gcc) [M db_mysql.so] km_dbase.o >>>> km_dbase.c:36:19: fatal error: mysql.h: No such file or directory >>>> #include <mysql.h> >>>> ^ >>>> compilation terminated. >>>> make[1]: *** [km_dbase.o] Error 1 >>>> make: *** [modules] Error 1 >>>> ------------------------------------------------------------------------ >>>> >>>> I checked and the mysql.h exists in the /usr/include/mysql directory. >>>> >>>> Any thoughts? >>>> >>>> FYI, here is my Dockerfile >>>> >>>> FROM centos:7 >>>> MAINTAINER “Sam D Ware” sware(a)o1.com >>>> ENV container docker >>>> RUN (cd /lib/systemd/system/sysinit.target.wants/; for i in ; do [ $i >>>> == systemd-tmpfiles-setup.service ] || rm -f $i; done); >>>> RUN rm -rf /lib/systemd/system/multi-user.target.wants/ \ >>>> && rm -rf /etc/systemd/system/.wants/ \ >>>> && rm -rf /lib/systemd/system/local-fs.target.wants/ \ >>>> && rm -f /lib/systemd/system/sockets.target.wants/udev \ >>>> && rm -f /lib/systemd/system/sockets.target.wants/initctl \ >>>> && rm -rf /lib/systemd/system/basic.target.wants/ \ >>>> && rm -f /lib/systemd/system/anaconda.target.wants/* >>>> VOLUME [ “/sys/fs/cgroup”] >>>> RUN yum -y install epel-release >>>> RUN yum -y update >>>> RUN yum -y install git-core gcc gcc-c++ flex bison mysql-devel >>>> RUN yum -y install mariadb-devel hiredis hiredis-devel curl nss make >>>> WORKDIR /usr/local/src/ >>>> RUN git clone --depth 1 --no-single-branch >>>> https://github.com/kamailio/kamailio kamailio >>>> WORKDIR /usr/local/src/kamailio/ >>>> RUN git checkout -b 5.3 origin/5.3 >>>> RUN make cfg >>>> COPY modules.lst /usr/local/src/kamailio/src/ >>>> RUN make all >>>> RUN make install >>>> RUN groupadd kamailio >>>> RUN adduser --system -g kamailio --shell /bin/false -c "Kamailio" >>>> --home /var/run/kamailio kamailio >>>> RUN cp /usr/local/src/kamailio/pkg/kamailio/obs/kamailio.sysconfig >>>> /etc/sysconfig/kamailio >>>> COPY kamailio.service /etc/systemd/system/kamailio.service >>>> RUN yum clean all; systemctl enable kamailio >>>> CMD ["/usr/sbin/init"] >>>> >>>> -- >>>> Sam D Ware >>>> >>>> >>>> _______________________________________________ >>>> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>> >>>> -- >>>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >>>> Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com >>>> >>>> _______________________________________________ >>>> Kamailio (SER) - Users Mailing List >>>> sr-users(a)lists.kamailio.org >>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing List >>> sr-users(a)lists.kamailio.org >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users(a)lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> > > > > ---------- Forwarded message ---------- > From: Sam Ware <sam.ware(a)gmail.com> > To: miconda(a)gmail.com > Cc: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> > Bcc: > Date: Fri, 10 Jan 2020 07:23:46 -0600 > Subject: Re: [SR-Users] Kamailio in Docker Compile Error > Thanks Daniel, > > That got me past that hurdle. It was the "which". Looks like the > pkg-config tools was already install; probably a preq for one of the other > pkgs. > > Sam > > On Fri, Jan 10, 2020 at 1:58 AM Daniel-Constantin Mierla < > miconda(a)gmail.com> wrote: > >> Hello, >> >> try to install which and pkg-config tools. >> >> Cheers, >> Daniel >> On 10.01.20 02:21, Sam Ware wrote: >> >> I am attempting to create my own docker image on Centos 7. During the >> build, I get the following error message during the "make all" phase. >> ---------------------------------------------------------------------- >> CC (gcc) [kamailio] core/cfg.tab.o >> LD (gcc) [kamailio] kamailio >> make[1]: which: Command not found >> make[1]: which: Command not found >> make[1]: which: Command not found >> make[1]: which: Command not found >> make[1]: which: Command not found >> make[1]: which: Command not found >> CC (gcc) [M db_mysql.so] km_dbase.o >> km_dbase.c:36:19: fatal error: mysql.h: No such file or directory >> #include <mysql.h> >> ^ >> compilation terminated. >> make[1]: *** [km_dbase.o] Error 1 >> make: *** [modules] Error 1 >> ------------------------------------------------------------------------ >> >> I checked and the mysql.h exists in the /usr/include/mysql directory. >> >> Any thoughts? >> >> FYI, here is my Dockerfile >> >> FROM centos:7 >> MAINTAINER “Sam D Ware” sware(a)o1.com >> ENV container docker >> RUN (cd /lib/systemd/system/sysinit.target.wants/; for i in ; do [ $i == >> systemd-tmpfiles-setup.service ] || rm -f $i; done); >> RUN rm -rf /lib/systemd/system/multi-user.target.wants/ \ >> && rm -rf /etc/systemd/system/.wants/ \ >> && rm -rf /lib/systemd/system/local-fs.target.wants/ \ >> && rm -f /lib/systemd/system/sockets.target.wants/udev \ >> && rm -f /lib/systemd/system/sockets.target.wants/initctl \ >> && rm -rf /lib/systemd/system/basic.target.wants/ \ >> && rm -f /lib/systemd/system/anaconda.target.wants/* >> VOLUME [ “/sys/fs/cgroup”] >> RUN yum -y install epel-release >> RUN yum -y update >> RUN yum -y install git-core gcc gcc-c++ flex bison mysql-devel >> RUN yum -y install mariadb-devel hiredis hiredis-devel curl nss make >> WORKDIR /usr/local/src/ >> RUN git clone --depth 1 --no-single-branch >> https://github.com/kamailio/kamailio kamailio >> WORKDIR /usr/local/src/kamailio/ >> RUN git checkout -b 5.3 origin/5.3 >> RUN make cfg >> COPY modules.lst /usr/local/src/kamailio/src/ >> RUN make all >> RUN make install >> RUN groupadd kamailio >> RUN adduser --system -g kamailio --shell /bin/false -c "Kamailio" --home >> /var/run/kamailio kamailio >> RUN cp /usr/local/src/kamailio/pkg/kamailio/obs/kamailio.sysconfig >> /etc/sysconfig/kamailio >> COPY kamailio.service /etc/systemd/system/kamailio.service >> RUN yum clean all; systemctl enable kamailio >> CMD ["/usr/sbin/init"] >> >> -- >> Sam D Ware >> >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> -- >> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >> Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com >> >> > > -- > Sam D Ware > > > > > ---------- Forwarded message ---------- > From: "Woscek, Martin W." <mwoscek(a)mitre.org> > To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> > Cc: > Bcc: > Date: Fri, 10 Jan 2020 16:57:29 +0000 > Subject: [SR-Users] Kamailio 5.2.2 IMS -enum_query question > > Hi, > > > > I am trying to invoke unum lookups from calls to enum_query() > ,is_from_user_enum() and enum_pv_query(). > > However we are not seeing any enum lookups over the wire using wireshark. > > > > In the Kamailio.cfg the following are set: > > > > loadmodule "enum" > > modparam("enum", "domain_suffix", "mydnsserver.org.") > > modparam("enum", "i_enum_suffix", " mydnsserver.org.") > > > > sample use in our Kamailio.cfg file: > > > > if(!enum_query("mydnsserver.org.")) { > > xlog(“….); > > #do something > > } else { > > Xlog(“….”) > > #do something else > > } > > > > I do see NAPTR lookups being done by the other Kamailio IMS processes > initially when it starts up so I know it has connectivity to the DNS > server, but none for the invoked enum queries. > > > > We are running Kamailio 5.2.2. > > > > Thanks, > > Martin > > > > ---------- Forwarded message ---------- > From: Sam Ware <sam.ware(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> > Cc: > Bcc: > Date: Fri, 10 Jan 2020 17:22:17 -0600 > Subject: [SR-Users] String Transformation Question > We are receiving traffic with a tech prefix. We want to be able to use a > # (hash tag) for the separator for the tech prefix and the remote user > ($rU). I was using the s.select transformation before when using * for > separator, but I get an error because it appears that it being interpreted > as a comment. Any suggestions? > > Example: > ## How I was doing it with the * > $avp(techprefix) = $(rU{s.select,0,*}); > > ## How I tried and got an error for the hash tag > $avp(techprefix) = $(rU{s.select,0,#}); > -- > Sam D Ware > > > > > ---------- Forwarded message ---------- > From: Alex Balashov <abalashov(a)evaristesys.com> > To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> > Cc: > Bcc: > Date: Fri, 10 Jan 2020 18:24:44 -0500 > Subject: Re: [SR-Users] String Transformation Question > What’s the error? > > — > Sent from mobile, with due apologies for brevity and errors. > > On Jan 10, 2020, at 6:23 PM, Sam Ware <sam.ware(a)gmail.com> wrote: > > > We are receiving traffic with a tech prefix. We want to be able to use a > # (hash tag) for the separator for the tech prefix and the remote user > ($rU). I was using the s.select transformation before when using * for > separator, but I get an error because it appears that it being interpreted > as a comment. Any suggestions? > > Example: > ## How I was doing it with the * > $avp(techprefix) = $(rU{s.select,0,*}); > > ## How I tried and got an error for the hash tag > $avp(techprefix) = $(rU{s.select,0,#}); > -- > Sam D Ware > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > > ---------- Forwarded message ---------- > From: David Villasmil <david.villasmil.work(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> > Cc: > Bcc: > Date: Fri, 10 Jan 2020 23:24:51 +0000 > Subject: Re: [SR-Users] String Transformation Question > Have you tried scaping it? Never done it, tho > > On Fri, 10 Jan 2020 at 23:23, Sam Ware <sam.ware(a)gmail.com> wrote: > >> We are receiving traffic with a tech prefix. We want to be able to use a >> # (hash tag) for the separator for the tech prefix and the remote user >> ($rU). I was using the s.select transformation before when using * for >> separator, but I get an error because it appears that it being interpreted >> as a comment. Any suggestions? >> >> Example: >> ## How I was doing it with the * >> $avp(techprefix) = $(rU{s.select,0,*}); >> >> ## How I tried and got an error for the hash tag >> $avp(techprefix) = $(rU{s.select,0,#}); >> -- >> Sam D Ware >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users(a)lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> > -- > Regards, > > David Villasmil > email: david.villasmil.work(a)gmail.com > phone: +34669448337 > _______________________________________________ > sr-users mailing list > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > -- Sam D Ware

3 3

Kamailio service error
by Ratanraj Singh 14 Jan '20

14 Jan '20

Hi All, I am troubling with kamailio service error. Can someone help on this issue? [image: image.png]

2 1

Trigger a custom SIP Notify Message on demand
by Rupesh Kumar 14 Jan '20

14 Jan '20

Hello kamailio gurus, I was wondering if there is a way to send a custom SIP NOTIFY message from kamailio to the registered SIP clients on-demand. For example, I want to take a kamailio node for maintainance and want all polycom phones registered on a kamailio node to re-register and move to a different kamailio node. I know the uac module can send custom requests via uac_req_send, but can I invoke this via a kamcmd or kamctl command ?Any pointers/clues would be a great help. Thanks,RK

2 3

topos and db_redis
by Ben Merrills 13 Jan '20

13 Jan '20

Hi All, Quick question, I've tried to find this myself, but there doesn't seem to be any documentation that I can find on it. Does the topos module work with db_redis module? I can see two schema files for topos (topos_t, topos_d), but I can't work out what the keys should be. modparam("db_redis", "keys", "topos_t=entry:hash_entry,hash_id,...?") modparam("db_redis", "keys", "topos_d=entry:hash_entry,hash_id,...?") modparam("topos", "db_url", "redis://127.0.0.1:6379/kamailio") modparam("topos", "storage", "db") Any pointers would be greatly appreciated. Kind Regards, Ben Merrills

2 1

Diameter EVENT_REQUEST
by Steve Bucklin 13 Jan '20

13 Jan '20

Dear All, I was wondering if someone could give me a pointer on handling diameter "EVENT REQUEST" (416) I am a complete newb to the diameter, and have used a RESTCOMM SMS-C to generate charging requests. 1. I have got the CDP and other modules going and it connects with application (Diameter Credit Control Application (4) 2. I tried to use a simple ims_ocs module, but get a "getMethod()" error. Looking at the code, there are only three supported, START, INTERIM, & STOP. 3. I have captured the message in pcap - and all looks OK. 4. To examine the whole message I tried using the ims_diameter_server module to produce a JSON of all the values sent in the $diameter_request within the event_route, HOWEVER once the request arrives I get an error: ERROR: cdp [diameter_avp.c:645]: AAAUngroupAVPS(): hss3g_ungroup_avps: source buffer to short!! Cannot read the whole AVP header! I assume that the cdp function AAAUngroupAVPS() cannot sort the data into JSON due to a buffer issue. Any pointers? Steve

1 0

Dúvidas
by Genildo Nascimento 13 Jan '20

13 Jan '20

Boa Noite. Instalei o kamailio mas não consigo usar Segue a configuração. baixei e adicionei chave GPG à sua lista de chaves apt wget -O- http://deb.kamailio.org/kamailiodebkey.gpg | sudo apt-key a Em seguida, adicionei as linhas do repositório ao seu arquivo /etc/apt/sources.list , Adicionei um repositório para a versão mais recente. sudo nano /etc/apt/source.list.d/kamailio.list versão ubuntu 16.04 deb http://deb.kamailio.org/kamailio53 xenial main deb-src http://deb.kamailio.org/kamailio53 xenial main Atualização do sistema e instalação do Kamailio sudo apt update sudo apt install kamailio kamailio-mysql-modules Editei o arquivo /etc/kamailio/kamctlrc e verifiquei se a DBENGINE variável está definida como MySQL. Removi o #símbolo para descomentá-lo. $ sudo nano / etc / kamailio / kamctlrc DBENGINE = MYSQL DBHOST = host local Instalei banco de dados apt install mysql-server modulos também apt -y install kamailio kamailio-mysql-modules kamailio-utils-modules rtpproxy Em seguida criei banco de dados para Mysql. sudo kamdbctl create logo após foi inserida a senha (xxxx) editei o arquivo /etc/kamailio/kamailio.cfg #! define WITH_MYSQL #! define WITH_AUTH *#! define WITH_USRLOCDB -> ao adicionar este iten o kamailio reiinicializa com erro (segue anexo arquivo jpeg)* #! define WITH_ACCDB Editei a linha da variável RUN_KAMAILIO no arquivo /etc/default/kamailio para que o kamailio possa rodar. RUN_KAMAILIO=yes configuração do rtpproxy no arquivo /ec/default/rtpproxy retirada do #simbolo control_sock=udp:localhost:7722 EXTRA_OPTS=" -l < IP local >" reinicio RTPproxy Obs: O sistema tambem não existe o arquivo /var/log/kamailio.log

1 0

String Transformation Question
by Sam Ware 10 Jan '20

10 Jan '20

We are receiving traffic with a tech prefix. We want to be able to use a # (hash tag) for the separator for the tech prefix and the remote user ($rU). I was using the s.select transformation before when using * for separator, but I get an error because it appears that it being interpreted as a comment. Any suggestions? Example: ## How I was doing it with the * $avp(techprefix) = $(rU{s.select,0,*}); ## How I tried and got an error for the hash tag $avp(techprefix) = $(rU{s.select,0,#}); -- Sam D Ware

3 2

Kamailio in Docker Compile Error
by Sam Ware 10 Jan '20

10 Jan '20

I am attempting to create my own docker image on Centos 7. During the build, I get the following error message during the "make all" phase. ---------------------------------------------------------------------- CC (gcc) [kamailio] core/cfg.tab.o LD (gcc) [kamailio] kamailio make[1]: which: Command not found make[1]: which: Command not found make[1]: which: Command not found make[1]: which: Command not found make[1]: which: Command not found make[1]: which: Command not found CC (gcc) [M db_mysql.so] km_dbase.o km_dbase.c:36:19: fatal error: mysql.h: No such file or directory #include <mysql.h> ^ compilation terminated. make[1]: *** [km_dbase.o] Error 1 make: *** [modules] Error 1 ------------------------------------------------------------------------ I checked and the mysql.h exists in the /usr/include/mysql directory. Any thoughts? FYI, here is my Dockerfile FROM centos:7 MAINTAINER “Sam D Ware” sware(a)o1.com ENV container docker RUN (cd /lib/systemd/system/sysinit.target.wants/; for i in ; do [ $i == systemd-tmpfiles-setup.service ] || rm -f $i; done); RUN rm -rf /lib/systemd/system/multi-user.target.wants/ \ && rm -rf /etc/systemd/system/.wants/ \ && rm -rf /lib/systemd/system/local-fs.target.wants/ \ && rm -f /lib/systemd/system/sockets.target.wants/udev \ && rm -f /lib/systemd/system/sockets.target.wants/initctl \ && rm -rf /lib/systemd/system/basic.target.wants/ \ && rm -f /lib/systemd/system/anaconda.target.wants/* VOLUME [ “/sys/fs/cgroup”] RUN yum -y install epel-release RUN yum -y update RUN yum -y install git-core gcc gcc-c++ flex bison mysql-devel RUN yum -y install mariadb-devel hiredis hiredis-devel curl nss make WORKDIR /usr/local/src/ RUN git clone --depth 1 --no-single-branch https://github.com/kamailio/kamailio kamailio WORKDIR /usr/local/src/kamailio/ RUN git checkout -b 5.3 origin/5.3 RUN make cfg COPY modules.lst /usr/local/src/kamailio/src/ RUN make all RUN make install RUN groupadd kamailio RUN adduser --system -g kamailio --shell /bin/false -c "Kamailio" --home /var/run/kamailio kamailio RUN cp /usr/local/src/kamailio/pkg/kamailio/obs/kamailio.sysconfig /etc/sysconfig/kamailio COPY kamailio.service /etc/systemd/system/kamailio.service RUN yum clean all; systemctl enable kamailio CMD ["/usr/sbin/init"] -- Sam D Ware

5 6

caching 404 responses
by Krzysztof Drewicz 10 Jan '20

10 Jan '20

Hello would like to see if I'm not reinventing the wheel: scenario: some old powerdialler/predictive calls kamailio as call dispatcher, kamailio then dials carrier. But this dialler does many calls to the same B number with 404 response (from carrier) in short (retry is < 600s) time, so no need to bother PSTN/carrier with retry. Should I simply use htable, cache 404 hit, use some reasonable value modparam("htable", "db_expires", 1) put there like 600 s TTL or something like that? Any hints/thoughts? Maybe this scenario is common and being used with success somewhere (and yes, this powerdialer is not very clever, and I do know it should not be needed for this) Thanks, -- *Krzysztof Drewicz* Senior Infrastructure Administrator CLUDO | ul. Grochowska 306/308, 03-840 Warszawa t+48221223977 kdrewicz(a)cludo.pl | www.cludo.pl <http://www.cludo.pl/>

2 1

TCP packet bundling
by Alex Balashov 09 Jan '20

09 Jan '20

Hi, I wanted to get clarity on a rather basic question: When using Kamailio with TCP, is it possible to end up with fragments of multiple SIP messages in a single "on-the-wire" packet? >From what I understand, the purpose of Kamailio's setting TCP_NODELAY is to prevent such bundling--as much as it is possible to control the fragmentation/packetisation behaviour of the transport protocol from a user-space application. -- 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/

3 7

Huge DB load while using location and registrar module (save()).
by Voip support 09 Jan '20

09 Jan '20

Dear Community, I am facing an issue with using the usrloc and registrar module to save registration informations in database. It works for me but when tested yesterday with around 1500-1600 users i got a lot of queries on database mostly SELECT. I figured out that changing db_mode to 0 resolves the issue because it works totally in memory (almost no select query on DB). So i think its location module. I tried db_mode=3 and db_mode=2 but both create a lot of load on database. The only function i need is to store some information of REGISTER in DB to know the count of registered users and the source IP / user-agent / username data. It needs to be saved in table on successfull REGISTRATION and removed when registration is expired or somebody UN-REGISTER. When used location module for that there was lots of queries like this (20000-50000 per second): select `contact`,`expires`,`q`,`callid`,`cseq`,`flags`,`cflags`,`user_agent`,`received`,`path`,`socket`,`methods`,`last_modified`,`ruid`,`instance`,`reg_id`,`server_id`,`connection_id`,`keepalive` from `location` where `username`='xxxx' order by q update `location` set `expires`='2020-01-08 18:51:39',`q`=-1.000000 ,`cseq`=2042,`flags`=0,`cflags`=64,`user_agent`='n/a',`received`='sip:xxxx:5060',`path`=NULL,`socket`='udp:xxxx:20051',`methods`=NULL,`last_modified`='2020-01-08 18:50:39',`callid`='1205705227@xxxx ',`instance`=NULL,`reg_id`=0,`server_id`=0,`connection_id`=-1,`keepalive`=1,`contact`='sip:1019@xxx:5060' where `ruid`='uloc-5e1642a3-2b90-0161' Looking up for ideas how to resolve this issue. I was thinking of implementing INSERT UPDATE and DELETE based on SQLops. But how to recognize that REGISTER was expiered? I would like location module to work mostly like in-memory. For each new registration INSERT entry to DB, each REGISTER re-new an UPDATE or DELETE should be triggered. So i would expect for 1000 users around 2000 operations per minute. Best regards, Tom

5 6

Kamailio v5.3.2 Released
by Daniel-Constantin Mierla 09 Jan '20

09 Jan '20

Hello, Kamailio SIP Server v5.3.2 stable release is out. This is a maintenance release of the latest stable branch, 5.3, that includes fixes since the release of v5.3.1. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.3.x. Deployments running previous v5.3.x versions are strongly recommended to be upgraded to v5.3.2. For more details about version 5.3.2 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/01/kamailio-v5-3-2-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! 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

1 0

Ask for Siremis document
by Thanh Tran 09 Jan '20

09 Jan '20

Hi, I looking for Administration Document for Siremis but can't find it. Can you share document ? Thanks,

2 1

Releasing Kamailio v5.3.2
by Daniel-Constantin Mierla 09 Jan '20

09 Jan '20

Hello, not so many fixes since the release of 5.3.1, but it looks enough to release a new minor version from branch 5.3. I am planning to do in on Thursday, Jan 9, 2020, if there are no new reports of relevant issues that make us to reconsider and postpone. Soon after we should probably do a release from branch 5.2 as well as the last one from branch 5.1. 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

1 1

Parsing exception when "user=phone" is present
by Duarte Rocha 08 Jan '20

08 Jan '20

Greetings, I'm testing some portability scenarios with carriers and i've ran into some issues. Take this example URI from ETSI TR 184 003 ( https://www.etsi.org/deliver/etsi_tr/184000_184099/184003/03.01.01_60/tr_18… ) - Sip:+49234598765;npdi;rn=+49-D123@provider5.net;user=phone In this case i need to parse user, user's parameters, domain and uri parameters. In order to that i'm using uri's transformations like {uri.user}, {uri.domain}, etc. However, when "user=phone" is present, Kamailio's parser works differently. Without "user=phone" -> {uri.user} is equal to "+49234598765;npdi;rn=+49-D123" With "user=phone" -> {uri.user} is equal to "+49234598765" . I really need to access "rn" and "npdi" parameters in those cases. How can i do it when "user=phone" is present? Is there a way to disable the "user=phone" exception for parsing? Best Regards, Duarte Rocha

2 2

possible TCP deadlock (tls again?) // pike module not releasing IPs
by Aymeric Moizard 08 Jan '20

08 Jan '20

Hi List, History: * In the past, I had deadlock which was, most probably, related to ssl1.1. We have discussed this issue, and a fix is supposed to workaround the issue that was detected. * With latest 5.2.X, I have experienced ONCE a similar behavior with TCP and TLS being mostly stuck. I have not been using this version much, but the fix was supposed to be in the core of kamailio. The status of the server this night: * I'm today running version: kamailio 5.3.1 (x86_64/linux), * Installed on stretch using http://deb.kamailio.org/kamailio53 repository. * This versions use libssl1.1 * A user reported that he can't connect with TCP * An average of 5000 IPs per 10 minutes are being banned by the pike module (could be twice the same) Yesterday/Today: * at the end of the outage, I had 2479 IP in my ipban htable. (which is equivalent to my statistics showing 2 bans/IP every 10 minutes = 5000) * looking at my logs, it appears that most (ALL?) ip being banned... are my regular users. * looking at my logs, I can't understand why pike would block them. This is a graph for statistics on my service for the last 24 hours: https://www.antisip.com/sip-antisip-com-register/status2.html Yesterday, at 22:18:39, kamailio started to BAN some IPs. 52 IPs were banned in a period of 10 minutes. I can confirm this from my logs. My pike configuration is this one: modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 64) modparam("pike", "remove_latency", 4) When detecting the issue, this morning, I typed: $> sudo kamctl stats $> sudo kamcmd htable.dump ipban //FAILURE (answer too large...) $> sudo kamctl trap Then, I started an agent with TCP and it worked...??? Then, a few seconds, may be a minute after: $> sudo kamcmd htable.dump ipban //SUCCESS and shows 2479 banned ip. and... everything is back to normal in a few minutes. I haven't restarted kamailio, and all statistics are as expected, as usual. Thus, it looks that " sudo kamctl trap" has triggered something. I already experienced a similar behavior -when testing my ssl1.1 deadlock last year-. 2 questions: 1/ I beleive my "pike" configuration should not ban users. Is my pike configuration wrong? As an example, pike has banned an IP sending one message/second. I believe my configuration should accept that? 2/ Could there still be a TLS issue with libssl1.1? This is the result of the "kamctl trap": https://sip.antisip.com/kamailio-pike-or-tls-issue-13-12-2019.kamctl-trap Sorry for the long story & hoping to find a long term solution or at least a workaround! Regards Aymeric -- Antisip - http://www.antisip.com

3 11

Re: [SR-Users] Add user issue - Multiicon
by technical＠multiicon.in 08 Jan '20

08 Jan '20

Dear Kamailio team, Greetings from Multiicon. We setuping kamailio sip server and all thing are done but when we going to add user we are getting Command not found. Using below command. I also check the server status by using systemctl status kamailio is showing active Please advise on this. Thanks & Regards, VIVEK AGHERA #technical department IND : +91 63548 23958 <mailto:vivek@multiicon.in> vivek(a)multiicon.in MULTIICON IDEOTECHNOLOGY PVT.LTD. 3rd Floor, Rumi Plaza, Airport Road, Rajkot - 360 001, Gujarat (INDIA) Please consider the environment before printing this e-mail.

2 1

WebRTC ACK Protocol
by Ilie Soltanici 08 Jan '20

08 Jan '20

Hello, I am trying to set up a WebRTC2SIP Gateway by using Kamailio and rtpengine. So far, everything is working fine, I'm able to register an extension and make a call, but for some reason, when i'm trying to call a WebRTC extension from any SIP Extension Kamailio is sending INVITE, WebRTC extension is sending back 200 OK, and then Kamailio is trying to send an ACK through UDP protocol, and not through wss, as it's supposed to do. This is how invite is looking: INVITE sip:nl7oe4ss@vjbh7r4im6j7.invalid;transport=wss SIP/2.0 Record-Route: <sip:my-company.net ;transport=udp;ftag=as1789445c;lr=on;nat=yes> Via: SIP/2.0/WSS 123.123.123.123:10443 ;branch=z9hG4bKe655.29d7c135a302f3eb803902d4f5a8da7e.0 Via: SIP/2.0/UDP 192.168.50.237:5060 ;received=192.168.50.237;branch=z9hG4bK7d2e534e;rport=5060 Max-Forwards: 70 From: "WebRTC" <sip:11@my-company.net>;tag=as1789445c To: <sip:15@192.168.50.210:5060> Contact: <sip:11@192.168.50.237:5060> Call-ID: 7fc800de060197fa2315c93763873092(a)my-company.net CSeq: 102 INVITE User-Agent: Proxy Date: Wed, 03 Apr 2019 17:11:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Alert-Info: Content-Type: application/sdp Content-Length: 596 Server: SIP Proxy and then WebRTC app is replying with 200 OK: SIP/2.0 200 OK Record-Route: <sip:my-company.net ;transport=udp;ftag=as1789445c;lr=on;nat=yes> Via: SIP/2.0/WSS 123.123.123.123:10443 ;branch=z9hG4bKe655.29d7c135a302f3eb803902d4f5a8da7e.0 Via: SIP/2.0/UDP 192.168.50.237:5060 ;received=192.168.50.237;branch=z9hG4bK7d2e534e;rport=5060 To: <sip:15@192.168.50.210:5060>;tag=dk4fa8ftt6 From: "WebRTC" <sip:11@my-company.net>;tag=as1789445c Call-ID: 7fc800de060197fa2315c93763873092(a)my-company.net CSeq: 102 INVITE Contact: <sip:nl7oe4ss@vjbh7r4im6j7.invalid;transport=wss> Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: outbound User-Agent: Proxy-WEBRTC Content-Type: application/sdp Content-Length: 901 and finally, Kamailio is trying to send this ack through UDP protocol: ACK sip:nl7oe4ss@22.22.22.22:57421;transport=wss SIP/2.0 Via: SIP/2.0/UDP 192.168.50.237:5060;branch=z9hG4bK56363ddf;rport Route: <sip:my-company.net;transport=udp;ftag=as1789445c;lr=on;nat=yes> Max-Forwards: 70 From: "WebRTC" <sip:11@my-company.net>;tag=as1789445c To: <sip:15@192.168.50.210:5060>;tag=dk4fa8ftt6 Contact: <sip:11@192.168.50.237:5060> Call-ID: 7fc800de060197fa2315c93763873092(a)my-company.net CSeq: 102 ACK User-Agent: Proxy Content-Length: 0 If i'm trying to force it through TLS, i'm receiving error: get_send_socket2(): protocol/port mismatch (forced tls:123.123.123.123:10443, to udp:22.22.22.22:23317) Can someone point me in the right direction, please? Thank you.

3 4

Kamailio when acting as client doesnt send SNI in client hello handshake message
by mahesh b 07 Jan '20

07 Jan '20

Hi, Am using Kamailio 5.1.9 version My Setup : client1 -> kamailio server 1 ( IP : 10.211.160.172) ----> kamailio server 2( IP : 10.211.160.176) -> client2 I have a scenario where kamailio server 1 has to initiate an outgoing tls connection to kamailio server 2, i have set the server_name and server_id in the client profile in tls.cfg like below on kamailio server 1 [client:default] verify_certificate = no require_certificate = no server_name = mahesh.client.com [client:10.211.160.172:5061] method = TLSv1+ verify_certificate = yes require_certificate = yes private_key = /root/mahesh_openssl/profile2/btip_172_server_private.key certificate = /root/mahesh_openssl/profile2/btip_172_server_public.crt ca_list = /root/mahesh_openssl/profile2/btip_ca_public.crt cipher_list = RSA verify_depth = 9 server_name = btip.176.com server_id = btip.176.com And in sar.cfg $xavp(tls=>server_name)="btip.176.com"; $xavp(tls=>server_id)="btip.176.com"; $du = "sip:10.211.160.176:5061;transport=tls"; .... t_relay(); What i observe is that , when client hello is sent by 10.211.160.172 to 10.211.160.176, i dont see Extension server_name being sent. Am i missing anything. Please help !

4 7

app_lua with kamailio 5.3.X
by Aymeric Moizard 07 Jan '20

07 Jan '20

Hi All! I have moved from kamailio 5.2.X to 5.3.X and I have now an error with calling "sr.err" from my lua script: sr.err("ERROR: param $tU [" .. tu .. "]\n") app_lua [app_lua_api.c:626]: app_lua_runstring(): error from Lua: /etc/kamailio/lua/my-script.lua:69: attempt to index global 'sr' (a nil value) I do understand that I should add this to continue with old way: loadmodule "app_lua_sr.so" modparam("app_lua_sr", "register", "sl") However, I would prefer to fix my script to use the NEW way but can't find it. Thanks a lot. Aymeric -- Antisip - http://www.antisip.com

2 1

RTPEngine performance - How far are you pushing your systems
by davy van de moere 07 Jan '20

07 Jan '20

Dear all, Not a direct question on kamailio, but on rtpengine, so feel free to ignore :) I have a farm of rtpengine servers , supporting our set of kamailio's. On recent decent hardware (hp dl360 gen9/10 servers), I typically push my systems upto 2500 channels per machine. Then I tend to see the load of the machine go up in a logarithmic manner. E.g. a load of 3 supports 2500 channels, but when its at 3000 channels, load can be at 8. etc. I have the kernel module running, and I have a good impression it's being used too. No transcoding is going on, no recording, ... I've tried on different OS'es, different network cards, and I always end up at that 2500 channels +-. Is that what I can expect from rtpengine? Or am I really overlooking something? What numbers do you push out of your systems? kind regards and happy 2020!

5 10

Siremis & freeswitch
by John Tuxies 07 Jan '20

07 Jan '20

I have a Debian server running Kamailio and i have all my users regiatering and talking to each other. I do create/remove users from Siremis. I have followed the guide from here: https://www.powerpbx.org/content/kamailio-v5-siremis-v5-debian-v9-mariadb-a… I would like to add freeswitch to the same server and add conference, voicemail, supplementary services for my users. I have given it a try to install freeswitch and tried to configure it accordingly to https://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc But.now the Siremis gui shows a lot of errors and i cannot login any more. Since it is a Virtual machine i reverted it back and it works without freeswitch. How could i integrate freeswitch to my system please?

4 11

opensource VoLTE/IMS Clients like IMSDroid
by Hamid Hashmi 06 Jan '20

06 Jan '20

Dear Users Can you please name some of VoLTE/IMS Clients (Like IMSDroid for Android) for Windows/Linux/MAC to test IMS Network. IMSdroid available clients on the internet are not working with android v8,9 and 10. or if someone can guide me to create a working APK from IMSDroid available code on Github<https://github.com/DoubangoTelecom/imsdroid>. Regards Hamid

1 0

SNI Query
by mahesh b 06 Jan '20

06 Jan '20

Hi All, Am using Kamailio 5.1.9 version. *Below is my tls.cfg* [server:default] method = TLSv1+ verify_certificate = no require_certificate = no private_key = server.key certificate = server.crt ca_list = bundle.crt cipher_list = RSA verify_depth = 9 [client:default] verify_certificate = no require_certificate = no [server:10.211.160.172:5061] method = TLSv1+ verify_certificate = yes require_certificate = yes private_key = /root/mahesh_openssl/profile2/btip_172_server_private.key certificate = /root/mahesh_openssl/profile2/btip_172_server_public.crt ca_list = /root/mahesh_openssl/profile2/btip_ca_public.crt cipher_list = RSA verify_depth = 9 server_name = btip.172.com [server:10.211.160.172:5061] method = TLSv1+ verify_certificate = yes require_certificate = yes private_key = /root/mahesh_openssl/profile1/ctip_172_server_private.key certificate = /root/mahesh_openssl/profile1/ctip_172_server_public.crt ca_list = /root/mahesh_openssl/profile1/ctip_ca_public.crt cipher_list = RSA verify_depth = 9 server_name = ctip.172.com My Kamailio server ip is 10.211.160.172 i)When i initiate a tls connection from remote server(which is also a kamailio server) say 10.211.160.176 to 10.211.160.172 In the client hello am setting sni name as btip.172.com => so on 10.211.160.172 side it is picking up the server profile with serve_name btip.172.com for the tls handshake.*// Working as expected* ii)When i initiate a tls connection from another remote server(Which is also a kamailio server) say 10.211.160.163 to 10.211.160.172 In the client hello am setting sni name as ctip.172.com => so on 10.211.160.172 side it is picking up the server profile with serve_name ctip.172.com for the tls handshake.*// Working as expected* iii)When i initiate a tls connection from another remote server(Which is also a kamailio server) say 10.211.160.175 to 10.211.160.172 In the client hello am NOT setting sni name => so on 10.211.160.172 side should it pick up the server default profile or the first profile to which IP and port matches ? what i observe from logs is that it is picking up the server profile with server_name ctip.172.com for the tls handshake. I had a look at the code in function tls_lookup_cfg, from the debug prints i understand it is trying to match profile for IP and port if ((p->port==0 || p->port == port) && ip_addr_cmp(&p->ip, ip))* // IP and port matched* { if(sname && sname->len>0) *//Incoming Client hello dint have sname, so it will hit the else part* { if(p->server_name.s && p->server_name.len==sname->len && strncasecmp(p->server_name.s, sname->s, sname->len)==0) { LM_DBG("socket+server_name based TLS server domain found\n"); return p; } } else { return p; *// so it is returning the first profile to which IP and port matched.* } } Am i missing anything or is this a bug ? if in the clienthello there is no sni , what needs to be done to make use of the default profile for the tls handshake ? Or is this something fixed in latest. I just Tried and Modified the code as below, after which it is giving the server default profile when no sni in Incoming Client Hello. if ((p->port==0 || p->port == port) && ip_addr_cmp(&p->ip, ip)) { if(sname && sname->len>0) { if(p->server_name.s && p->server_name.len==sname->len && strncasecmp(p->server_name.s, sname->s, sname->len)==0) { LM_DBG("socket+server_name based TLS server domain found\n"); return p; } } else { if( (type & TLS_DOMAIN_SRV) && (p->server_name.s) ) { LM_DBG("Inside %s at %d\n",__FUNCTION__,__LINE__); return cfg->srv_default; } else { LM_DBG("Inside %s at %d\n",__FUNCTION__,__LINE__); return p; } } } Regards, Mahesh.B

2 4

Installation issue
by Hassen Bencheikh 06 Jan '20

06 Jan '20

Good morning, I am a telecommunication engineer from the National Establishment of Aerial Navigation (Algeria). First of all i would like to thank you for making the SIP sever Kamailio available. But actually i have got issues during the installation process. I was using the steps described on the link below : https://www.kamailio.org/wiki/install/devel/git The issue was encountered at the step #make all, an error was found, and here is what was displayed on the terminal: *linux-vd3w:/usr/local/src/kamailio-devel/kamailio # make allmake -C src/ all make[1]: Entering directory '/usr/local/src/kamailio-devel/kamailio/src'generating core/autover.h ...CC (gcc) [kamailio] core/ver.oLD (gcc) [kamailio] kamailiowhich: no mysql_config in (/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin)which: no mysql_config5 in (/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin)which: no mariadb_config in (/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin)which: no mysql_config in (/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin)which: no mysql_config5 in (/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin)which: no mariadb_config in (/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin)CC (gcc) [M db_mysql.so] my_res.oIn file included from my_res.c:24:0:my_cmd.h:27:10: fatal error: mysql.h: No such file or directory #include <mysql.h> ^~~~~~~~~compilation terminated.make[2]: *** [../../Makefile.rules:100: my_res.o] Error 1make[1]: *** [Makefile:511: modules] Error 1make[1]: Leaving directory '/usr/local/src/kamailio-devel/kamailio/src'make: *** [Makefile:30: all] Error 2linux-vd3w:/usr/local/src/kamailio-devel/kamailio #* Actually my knowledge on linux (opensuse leap) that i am using is very limited. Hopefully i will be able to use the server once installed and it will be used for ip recording tests purpose. Thank you for reading this email. Good bye.

5 4

MSRP proxy
by Sergey Safarov 06 Jan '20

06 Jan '20

I want proxy MSRP data on kamailio for SIP call (not websocket). To do this need update connection information for "message" media in SDP. How i can properly do it only for one media (other medias will be proxed using rtpproxy or rtpengine ). Looks as fix_nated_sdp fix for all medias in SDP. Sergey

2 1

pv_parse_spec2(): wrong char [)/41] in [$T(reply_reason)] at [15 (5)]
by Denys Pozniak 06 Jan '20

06 Jan '20

Hello! Looks like $T(reply_reason) does not work well (although $T_reply_reason is ok). Kamailio script: failure_route[TRUNK_FAILURE] { ... xlog("L_WARN","rm=$rm / $T(reply_reason) \n"); ... } 0(8894) ERROR: tmx [t_var.c:696]: pv_parse_t_name(): unknown PV name reply_reason 0(8894) ERROR: <core> [core/pvapi.c:952]: pv_parse_spec2(): pvar "T" has an invalid name param [reply_reason] 0(8894) ERROR: <core> [core/pvapi.c:1107]: pv_parse_spec2(): wrong char [)/41] in [$T(reply_reason)] at [15 (5)] 0(8894) ERROR: xlog [xlog.c:511]: xdbg_fixup_helper(): wrong format[rm=$rm / $T(reply_reason) https://www.kamailio.org/wiki/cookbooks/5.2.x/pseudovariables $T(name) - pseudo-variable class to access TM attributes The *name* can be: - reply_reason - reply reason [kamailio]$ kamailio -v version: kamailio 5.2.2 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, 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_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 18:04:26 Apr 3 2019 with gcc 4.8.5 -- BR, Denys Pozniak

2 1

Tweaking TCP settings
by Dovid Bender 06 Jan '20

06 Jan '20

Hi, First off please forgive my lack of knowledge on how TCP works. We are using Kamailio 5.0.7 and we have an issue where clients are connecting via TCP and their NAT devices are closing up. Because of this we want to send TCP keep alives every so often. 1) When restarting Kamailio it's sends a RST. Is this Kamailio sending it out or is it linux sending it when the application is killed? The issue we have is if say we need to do a restart 2-3 times (yes we should normally do that) then we end up with 3x registrations in the db (since when using db mode if the connection goes away it won't remove the reg from the db (as per - https://kamailio.org/docs/modules/5.0.x/modules/usrloc.html#usrloc.p.handle… ) 2) I haven testing with the following settings. a. tcp_crlf_ping=yes b. tcp_keepcnt = 3 c. tcp_keepidle = 5 With the above I see the TCP keep alives coming in every 75 seconds. If I tcp_keepintvl = 10 then I see TCP keep alives going out from Kamailio to the phone. As per https://www.kamailio.org/wiki/cookbooks/5.0.x/core#tcp_keepintvl it says "Time interval between keepalive probes, when the previous probe failed". Looking at my captures Kamilio sends out the TCP keep alive and gets it back. Is Kamailio not seeing it? With the above it seems to be working the way I want it but I want to make sure that I am doing it right. 3) For the devs on here how hard would it be to implement handle_lost_tcp for DB-Only? TIA and a happy new year to all. Regards, Dovid

2 1

KSR.sdpops.remove_media is not working as expected
by Miteshkumar Thakkar 06 Jan '20

06 Jan '20

Hi, I wants to remove media type "image" from SDP. For that I have tried to use KSR.sdpops.remove_media but it is not working. Not working in the sense that there is no effect of that function on SDP. I am using lua. Kindly find more details below. *Kamailio version*: kamailio 5.2.5 (x86_64/linux) 62d35f *SDP to process:* INVITE sip:xxxxxxxxxx@xxxxxxxxx.com:5060 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK-60823-1-0 From: xxxxxxxxxx <sip: xxxxxxxxxx@ xxxxxxxxxx.com:5060>;tag=xxxxx To: xxxxxxxxxx <sip: xxxxxxxxxx@ xxxxxxxxxx.com:5060> Call-ID: 1-60823(a)172.16.19.64 CSeq: 1 INVITE Contact: sip:xx@xxxxxxx:5060 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 337 v=0 o=zt 53655765 2353687637 IN IP4 xxx.xxx.xxx.xxx s=- c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=image xxxx RTP/AVP udptl t38 a=sendrecv a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:262 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy m=audio xxxx RTP/AVP 0 a=rtpmap:0 PCMU/8000 *Code snippet:* KSR.sdpops.remove_media("image") logger.log("info", "SDP:" .. headers.get("$sdp(body)")) *Log snippet:* https://pastebin.com/5JZmQUfq One more thing I would like to mention that if I do not use dialog module then the function KSR.sdpops.remove_media works. But I can not avoid using dialog module. Thanks in advance. Mitesh

3 8

dlg_set_timeout error
by Pepelux 03 Jan '20

03 Jan '20

Hi all I'm trying to set a dialog timeout when a call is established: event_route[dialog:start] { if(dlg_set_timeout("10")) { xinfo("10 secs"); } } But when the time expired I obtain this error: Jan 3 11:24:20 test /usr/sbin/kamailio[18436]: CRITICAL: dialog [dlg_timer.c:199]: update_dlg_timer(): Trying to update a bogus dlg tl=0x93947b30 tl->next=(nil) tl->prev=(nil) Jan 3 11:24:20 test /usr/sbin/kamailio[18436]: ERROR: dialog [dlg_hash.c:1267]: update_dlg_timeout(): failed to update dialog lifetime # kamailio -v version: kamailio 5.3.1 (i386/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, 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_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 6.3.0 Any idea? Thanks and Happy New Year

3 6

rtpproxy does not response, disable it
by Zodiac 03 Jan '20

03 Jan '20

Dear friends, I am working on a program on Kamailio and rtpengine proxy. I am wondering whether can I set Kamailio and rtpengine daemon on different physical machines. For example, I set Kamailio on a machine with IP address:10.109.247.80, and launch rtpengine daemon on another machine with interface parameter as 10.109.247.90 and ng port 7723. I set parameter in Kamailio.cfg with modparam(“rtpengine”, “rtpengine_sock”, “udp:10.109.247.90:7723”). Unfortunately I got debug message like this: ERROR: rtpengine [rtpengine.c:1710]: send_rtpp_command(): can't send command to a RTP proxy ERROR: rtpengine [rtpengine.c:1746]: send_rtpp_command(): proxy <udp:10.109.247.90:7723> does not respond, disable it ERROR: rtpengine [rtpengine.c:1616]: rtpp_test(): proxy did not respond to ping And, I also tried to set Kamailio and rtpengine daemon in a same machine,and use modparam(“rtpengine”, “rtpengine_sock”, “udp:localhost:7723”). And Kamailio can work functionally under this situation. rtpengine daemon can receive ping message from Kamailio and rtpengine daemon can work as suspected. So for the later case, is it supposed that Kamailio be in the same machine with same localhost address? Otherwise, what’s the reason for my ERROR? ------------------------------------ 北京邮电大学网络技术研究院网络与交换技术国家重点实验室田军 +86 18810315790 mozillafire(a)bupt.edu.cn ------------------------------------

5 10

rtjson and uac_auth not updating to/from headers
by Ben Merrills 02 Jan '20

02 Jan '20

Hi, I am using rtjson module to do some call routing, as part of that I've added an additional section to the document to include authentication information. I am using the uac_auth function from within failure_route to send authentication on to the uri. This works fine... however. When the initial invite is sent, the To/From headers are updated as per the rtjson document. This isn't the case however when the uac_auth function re-sends the invite with the digest header. The original to/from are being sent. I am using the dialog module restore_mode = auto restore_dlg = 1 I assume there is something I need to call, but none of the RTJSON functions seem to be available within failure_route. Any pointers as to where I am going wrong would be greatly appreciated. Thanks in advance. Ben failure_route[TRUNKAUTH] { if (t_is_canceled()) { exit; } if(t_check_status("401|407")) { $avp(auser) = $avp(s:authUser); $avp(apass) = $avp(s:authPass); uac_auth(); t_relay(); exit; } }

2 2

Changing name of the 'versions' table
by Ben Kaufman 31 Dec '19

31 Dec '19

Is it possible to use a table other than 'version' a the version table? i.e. if I'm in a shared DB, and there's an existing table named 'version' with completely different schema, is there a way I can create a table like 'kamailio_version' with the correct schema and data and use it instead? -- Ben Kaufman STATEMENT OF CONFIDENTIALITY: The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, or the person responsible for delivering the e-mail to the intended recipient, be advised you have received this message in error and that any use, dissemination, forwarding, printing, or copying is strictly prohibited. Please notify AltiGen Communications immediately at either (888)258-4436 or via email to administrator(a)altigen.com, and destroy all copies of this message and any attachments.

3 2

Calculate call duration
by Duarte Rocha 31 Dec '19

31 Dec '19

Greetings, How sould a call duration be calculated? Let's say the call creation on Kamailio as a proxy has those steps : 1 - Invite is received 2 - Provisional responses 3 - 200 OK is received 4 - ACK to 200 OK is received 5 - BYE is received 6 - 200 OK to BYE is received Should the duration begin to count on step 3 (200-Ok Received) or 4 (ACK received) ? Should it end on the step 5 or 6? ACC module tells me that start time for CDR can either be on dialog creation or confirmation - https://www.kamailio.org/docs/modules/5.2.x/modules/acc.html#acc.p.cdr_star… - What is the dialog creation ? Is it when the INVITE is received ? - According to DIALOG module, a confirmed dialog can be "Waiting for ACK" or "Active Call". Which one is used by acc module? Best Regards,

3 2

Merry Christmas and Happy Holidays!
by Daniel-Constantin Mierla 25 Dec '19

25 Dec '19

Another year getting to its end, the season holidays are ahead, therefore it is no better time than now to express my thanks and greetings to all the friends, developers, supporting companies and community members that made 2019 an outstanding year for Kamailio project. Enjoy the holidays! Merry Christmas! Daniel * Santa is flying Kamailio! * -- 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

4 3

topos: how to hide one side only
by Dmitri Savolainen 24 Dec '19

24 Dec '19

Hi all. I have Kamailio with 2 addr: kamailio_external_ip - for client interconnect kamailio_internal_ip - for local media server interconnect Like this: client - kamailio_external_ip - kamailio_internal_ip - media_server "topos" module hide sip headers for both "client" and "media_server" side. Is it true: with topos i can't hide headers only for the "client" side and show them for the "media_server"? I want full header set visible for the "media_server", but only kamailio_external_ip visible for the "client". -- Savolainen Dmitri

2 6

Getting error in auth modules- Installing Kamalio 5.2
by Mahesh Kumar 24 Dec '19

24 Dec '19

Hi Team, Getting error in auth modules during Kamalio 5.2 installation and doing SIP routing, anyone know this why i am getting below error..?? Please support..!! Error Snapshot ======================================== (2807) ERROR: <core> [core/cfg.y:3370]: yyparse(): cfg. parser: failed to find command www_authorize (params 2) 0(2807) CRITICAL: <core> [core/cfg.y:3510]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 1081, column 48: unknown command, missing loadmodule? 0(2807) ERROR: <core> [core/cfg.y:3370]: yyparse(): cfg. parser: failed to find command www_challenge (params 2) 0(2807) CRITICAL: <core> [core/cfg.y:3510]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 1082, column 29: unknown command, missing loadmodule? 0(2807) ERROR: <core> [core/cfg.y:3370]: yyparse(): cfg. parser: failed to find command consume_credentials (params 0) 0(2807) CRITICAL: <core> [core/cfg.y:3510]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 1086, column 22: unknown command, missing loadmodule? 0(2807) ERROR: <core> [core/cfg.y:3370]: yyparse(): cfg. parser: failed to find command proxy_authorize (params 2) 0(2807) CRITICAL: <core> [core/cfg.y:3510]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 1097, column 41: unknown command, missing loadmodule? 0(2807) ERROR: <core> [core/cfg.y:3370]: yyparse(): cfg. parser: failed to find command proxy_challenge (params 2) 0(2807) CRITICAL: <core> [core/cfg.y:3510]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 1098, column 31: unknown command, missing loadmodule? 0(2807) ERROR: <core> [core/cfg.y:3370]: yyparse(): cfg. parser: failed to find command consume_credentials (params 0) 0(2807) CRITICAL: <core> [core/cfg.y:3510]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 1102, column 22: unknown command, missing loadmodule? ======================================== Commands in Kamalio.cfg =================================================== route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!www_authorize("192.168.1.67","subscriber")) { www_challenge("$td","0"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("$fd","subscriber")) { proxy_challenge("$fd","0"); break; }; consume_credentials(); # Forward to SDIN S3 rewritehostport("uk.sdin.bt.net:5060"); forward(uk.sdin.bt.net,5060); break; } route[4] { # ----------------------------------------------------------------- # CANCEL or ACK Message Handler # ----------------------------------------------------------------- # Forward to SDIN S3 rewritehostport("uk.sdin.bt.net:5060"); forward(uk.sdin.bt.net,5060); break; } ========================================================

2 1

Kamailio at 36c3, Happy holidays
by Henning Westerholt 24 Dec '19

24 Dec '19

Hello, I will be at the 36c3 [1] congress in Leipzig. So, if you are also there and want to chat about Kamailio or other RTC topics, please contact me. I wish you all Merry Christmas and a Happy New Year! Henning [1] https://events.ccc.de/congress/2019/wiki/index.php/Main_Page -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

1 0

Not working : Using $cfg(line) to print line number
by mahesh b 24 Dec '19

24 Dec '19

Hi All, Below is the syntax am using to print the line number in kamailio configuration file. xlog("L_WARN","Hi Mahesh at line $cfg(line)\n"); Unfortunately , when i do this looks like kamailio is not starting, Any help please regarding what am i missing here. I tried Xlogl as well, which doesnt print the line number by default in my setup. This is the details i had read to understand config file attributes : $cfg(key) - Config File Attributes Attributes related to configuration file. The key can be: - line - return the current line in config - name - return the name of current config file - file - return the name of current config file - route - return the name of routing block Example: send_reply("404", "Not found at line $cfg(line)")

2 2

rtpproxy_manage() doesn't recognize SDP when Content-Length is not present
by Leonid Fainshtein 22 Dec '19

22 Dec '19

Hi, According to RFC-3261 p.20.14, the Content-Length header field must be present only if a stream-base transport protocol (such as TCP) is used. Therefore, in the case of the UDP transport protocol, this field is optional. Unfortunately, the rtpproxy_manage() function in Kamailio v.5.2.5 doesn't recognize the SDP body when the Content-Length header field is not present: Dec 22 08:58:34 pbx kamailio[13456]: 4(13488) WARNING: {1 1 INVITE wrjxwyuke5ukvvwwxd00zwtu0ktfuzuf(a)10.18.5.64} sanity [sanity.c:585]: check_cl(): content length header missing in request Dec 22 08:58:34 pbx kamailio[13456]: 4(13488) INFO: {1 1 INVITE wrjxwyuke5ukvvwwxd00zwtu0ktfuzuf(a)10.18.5.64} <script>: rtproxy ei. $source:81.xxx.xxx.xxx, INVITE sip:098232787@xxx.xxx.xxx.xxx:5060;user=phone $ Dec 22 08:58:34 pbx kamailio[13456]: 4(13488) ERROR: {1 1 INVITE wrjxwyuke5ukvvwwxd00zwtu0ktfuzuf(a)10.18.5.64} rtpproxy [rtpproxy_funcs.c:182]: extract_body(): failed to get the content length in message Dec 22 08:58:34 pbx kamailio[13456]: 4(13488) ERROR: {1 1 INVITE wrjxwyuke5ukvvwwxd00zwtu0ktfuzuf(a)10.18.5.64} rtpproxy [rtpproxy.c:2271]: force_rtp_proxy(): can't extract body from the message Dec 22 08:58:34 pbx rtpproxy[1386]: DBUG:GLOBAL:get_command: received command "13498_4 LSEIc0,101 wrjxwyuke5ukvvwwxd00zwtu0ktfuzuf(a)10.18.5.64 10.70.227.183 13592 d5k0w3ft-CC-1116-OFC-253;1 as0bf73ed7;1" Dec 22 08:58:34 pbx rtpproxy[1386]: INFO:GLOBAL:handle_command: lookup request failed: session wrjxwyuke5ukvvwwxd00zwtu0ktfuzuf(a)10.18.5.64, tags d5k0w3ft-CC-1116-OFC-253;1/as0bf73ed7;1 not found Dec 22 08:58:34 pbx rtpproxy[1386]: DBUG:GLOBAL:rtpc_doreply: sending reply "0\n" Also, the textops' has_body() function always checks the “Content-Length” presence that is not 100% correct behavior. Just for your information, the problem was discovered in a system where Huawei SoftX3000 is used. It looks like the softswitch removes the “Content-Length” field in an attempt to decrease size of SIP message when it is too long. If the user disables some codecs in their SIP phone then SoftX3000 keeps the “Content-Length” untouched. Should I open a bug regarding this matter? Best regards, Leonid Fainshtein Xorcom Ltd

2 1

Disabling weak SSL Cypher suites
by Arik Halperin 22 Dec '19

22 Dec '19

Hello, How can I disable: TLS_RSA_WITH_RC4_128_SHA (0x5) INSECURE128 TLS_RSA_WITH_RC4_128_MD5 (0x4) INSECURE128 What should I put in cypher_list in order to disable the above? I would also like support TLS 1.2 and TLS 1.3, but remove support for 1.0 and 1.1 Thanks, Arik Halperin

3 5

Need support
by Srikanth Magloori 21 Dec '19

21 Dec '19

Hi Team, We are using your product Kamalio in Call Recording services. We are planning for Performance testing of our recording system. Currently we have bypassed Twilio and Kamailio. Generating Calls using SIPP directly to VOIP. So, we want calls to route through KAMAILIO -> VOIP. So performance can be bit realistic. Any help is greatly appreciated! Thanks Srikanth

4 3

Questions about Presence
by Володимир Іванець 20 Dec '19

20 Dec '19

Hello! I'm testing Kamailio as a presence server with configuration suggested by https://kb.asipto.com/kamailio:presence:k43-blf tutorial. When I set presence module subs_db_mode setting to 0 or 2 (did not test with 1 and 3 seemed to have no problems) I was getting "DEBUG: presence [notify.c:1234]: publ_notify(): Could not find subs_dialog" message after Kamailio received some PUBLISH requests. And NOTIFY is not being generated. It appears to be a mistake. There should be an active watcher according to Expires header values of the previous SUBSCRIBE request and the response to it. I could not find how to get the list of active watchers from Kamailio memory. *Is it possible or it can be done only by adding extra debug line in get_subs_dialog function?* Also I found that after setting pua module db_mode setting to 0, Kamailio is still writing records to presentity table. And it is constantly queried "select `username`,`domain`,`etag`,`event` from `presentity` where `expires`<1576584910 AND `expires`>0 order by username" during the run time. *Is it correct behavior?* Thank a lot!

2 2

Kamailio Product Support
by Simal Mathai 20 Dec '19

20 Dec '19

Hi Team, We are deploying Kamailio 5.3.1 on CentOS 7 for testing with a known carrier in Europe. We would like to get support for the product installation. Regards, Simal Mathai O: +1 214 242 5975 [cid:image001.png@01D13657.67E2B360]<http://www.tekvizion.com/> Please follow us on LinkedIn and Twitter [cid:image002.png@01D13657.67E2B360]<https://www.linkedin.com/company/tekvizion?trk=top_nav_home>[cid:image003.png@01D13657.67E2B360]<https://twitter.com/tekvizion>

2 1

how to detect "failure to add branches"?
by Juha Heinanen 20 Dec '19

20 Dec '19

In route block I call t_on_failure, t_on_branch_failure, and t_on_branch. If branch route fails like this: Dec 15 08:52:19 box /usr/bin/sip-proxy[84346]: ERROR: tm [ut.h:284]: uri2dst2(): failed to resolve "foo.tutpro.com" :unresolvable A or AAAA request (-7) Dec 15 08:52:19 box /usr/bin/sip-proxy[84346]: ERROR: tm [t_fwd.c:1738]: t_forward_nonack(): failure to add branches failure_route or tm:branch-failure event_route is not executed. Is there a way to catch such a failure? -- Juha

3 5

TLS testing (with sipp)
by Sebastian Damm 19 Dec '19

19 Dec '19

Hi, I'm trying to construct an end-to-end encrypted signalling test through our setup. I thought I could use sipp for that, as it supports TLS according to the man page. However, when I try to run it, I get this error: FI_init_ssl_context: SSL_CTX_use_certificate_file failed. I searched the web; however, all similar questions end up without answers. I tried specifying a local key and cert without success. I'd think I should not need a cert for my client, though. Has anyone ever successfully conducted an automated TLS test? I'm open to using a different tool if necessary. Thanks for all hints or examples. Regards, Sebastian -- Sebastian Damm Voice Engineer __________________________________________ sipgate GmbH Gladbacher Straße 74 | 40219 Düsseldorf

4 6

Kamailio SIP TLS Troubleshooting
by Sudhakar Parasher 19 Dec '19

19 Dec '19

Hi Experts. Need your opinion on below part related to kamailio I am using SIP Kamailio module Version for the SIP Registrations from the Client mobile phones using Android and IOS which is Over TLSv1.2 , So in nutshell the SIP Signaling is TLS version 1.2 encrypted and Customer is requesting for the forward of the unencrypted SIP signaling messages especially the SIP Registration messages to their probe solutions without storing the SIP messages locally in the Kamailio Database Question :Would it not be possible for Kamailio imply “mirror” all the SIP messages related to all SIP REGISTER dialogs (after removing TLS for inbound messages and prior to adding TLS for outbound message) to a designated IP Address and Port? No need for local database storage. No need for encapsulation. No mirroring of other messages/dialogs. Cheers Sudhakar

4 10

[dispatcher] freeing already freed pointer, called from core: core/xavp.c: xavp_free
by Denys Pozniak 19 Dec '19

19 Dec '19

Hello! The configuration below leads to a CRITICAL message in the console. Please let me know how to avoid this issue. event_route[htable:mod-init] { route(AUTORUN); ... } route[AUTORUN] { if ( ds_select("9", "4") ) { xlog("L_WARN", "[$cfg(route)] *ds_select* $xavp(_dsdst_[0]=>uri) $xavp(_dsdst_[1]=>uri) $xavp(_dsdst_[2]=>uri)\n"); } ... } dispatcher.list # group 9 9 sip:192.168.20.125:5070 9 sip:192.168.20.160:5070 9 sip:192.168.20.7:5070 9 sip:192.168.20.107:5070 9 sip:192.168.20.190:5070 0(6088) DEBUG: dispatcher [dispatch.c:2054]: ds_manage_routes(): set [9] 0(6088) DEBUG: dispatcher [dispatch.c:2158]: ds_manage_routes(): using alg [4] hash [0] 0(6088) DEBUG: dispatcher [dispatch.c:2199]: ds_manage_routes(): selected [4-9-2/0] <sip:192.168.20.190:5070> 0(6088) DEBUG: dispatcher [dispatch.c:2218]: ds_manage_routes(): using first entry [9/0] 0(6088) DEBUG: dispatcher [dispatch.c:2238]: ds_manage_routes(): using entry [9/1] 0(6088) DEBUG: dispatcher [dispatch.c:2238]: ds_manage_routes(): using entry [9/2] 0(6088) DEBUG: dispatcher [dispatch.c:2238]: ds_manage_routes(): using entry [9/3] 0(6088) DEBUG: dispatcher [dispatch.c:2272]: ds_manage_routes(): using default entry [9/4] 0(6088) DEBUG: dispatcher [dispatch.c:2016]: ds_select_dst_limit(): selected target destinations: 5 0(6088) WARNING: <script>: [AUTORUN] *ds_select* sip:192.168.20.190:5070 sip:192.168.20.107:5070 sip:192.168.20.7:5070 1(6089) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list 0x7fe5f7d4f420 1(6089) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(6089) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(6089) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(6089) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(6089) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(6089) DEBUG: <core> [core/xavp.c:495]: xavp_destroy_list(): destroying xavp list 0x7fe5f7d4faf0 1(6089) DEBUG: <core> [core/xavp.c:495]: xavp_destroy_list(): destroying xavp list (nil) 1(6089) CRITICAL: <core> [core/mem/q_malloc.c:514]: qm_free(): BUG: freeing already freed pointer (0x7fe5f7d4faf0), called from core: core/xavp.c: xavp_free(65), first free core: core/xavp.c: xavp_free(65) - ignoring 1(6089) DEBUG: <core> [core/xavp.c:495]: xavp_destroy_list(): destroying xavp list (nil) 1(6089) CRITICAL: <core> [core/mem/q_malloc.c:514]: qm_free(): BUG: freeing already freed pointer (0x7fe5f7d4f168), called from core: core/xavp.c: xavp_free(65), first free core: core/xavp.c: xavp_free(65) - ignoring 1(6089) DEBUG: <core> [core/xavp.c:495]: xavp_destroy_list(): destroying xavp list (nil) 1(6089) CRITICAL: <core> [core/mem/q_malloc.c:514]: qm_free(): BUG: freeing already freed pointer (0x7fe5f7d4f378), called from core: core/xavp.c: xavp_free(65), first free core: core/xavp.c: xavp_free(65) - ignoring 1(6089) DEBUG: <core> [core/xavp.c:495]: xavp_destroy_list(): destroying xavp list (nil) 1(6089) CRITICAL: <core> [core/mem/q_malloc.c:514]: qm_free(): BUG: freeing already freed pointer (0x7fe5f7d4f580), called from core: core/xavp.c: xavp_free(65), first free core: core/xavp.c: xavp_free(65) - ignoring 1(6089) DEBUG: <core> [core/xavp.c:495]: xavp_destroy_list(): destroying xavp list (nil) 1(6089) CRITICAL: <core> [core/mem/q_malloc.c:514]: qm_free(): BUG: freeing already freed pointer (0x7fe5f7d4f790), called from core: core/xavp.c: xavp_free(65), first free core: core/xavp.c: xavp_free(65) - ignoring 1(6089) DEBUG: <core> [core/xavp.c:495]: xavp_destroy_list(): destroying xavp list (nil) 1(6089) CRITICAL: <core> [core/mem/q_malloc.c:514]: qm_free(): BUG: freeing already freed pointer (0x7fe5f7d4f9a0), called from core: core/xavp.c: xavp_free(65), first free core: core/xavp.c: xavp_free(65) - ignoring 1(6089) DEBUG: <core> [core/receive.c:457]: receive_msg(): cleaning up ... version: kamailio 5.2.2 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, 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_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 19:29:40 Jun 20 2019 with gcc 4.8.5 -- BR, Denys Pozniak

2 2

kamailio 5.2.5 wrong send socket on a mhomed proxy.
by Vitalii Aleksandrov 19 Dec '19

19 Dec '19

Hi, kamailio 5.2 branch has a commit 09ac3e47ef79dbd599b7dec5b84ae3b792f025ed cherry picked from master branch. It makes sense for master because get_send_socket2() behavior has changed and function became started and also there were some patches to propagate connection it to uac structure. 5.2 kamailio takes the first interface for TCP/TLS/WS connections and then finds existing tcp connection matched by ip:port, but when the patch is applied connection lookup fails and kamailio can't forward in-dialog messages.

2 1

How to reduce re-INVITE time in Kamailio
by Anuran Barman 18 Dec '19

18 Dec '19

Hi, I am using Kamailio for my SIP Phone app. My main SIP client is the iOS app. Till now everything is working fine regarding the calling. But the problem is when the app is killed I am not able to pick up incoming call. The thing is, when app is killed I am sending PushKit VOIP message to my ios app to wake it up. App is waking up. But by the time app is waking up, the original INVITE message got missed up. So according to SIP protocol it will re-INVITE after some time. Currently if I keep my app open for 10-15 secs after it wakes up, I am able to get the re-INVITE and everything works fine just as it was supposed to. But waiting 10-15 sec on the app is not practical in any sense. So my question is there any way I can reduce the re-INVITE time to like 5/6 seconds so that as soon as the app open it will wait for 5/6 seconds (practical upto some extent) and it will get the call ? Or what kamailio is using as the interval time is a SIP standard and can not be changed? If possible to do so, how can I do that? Please help.

6 11

kamcmd stuck
by Andrew Chen 17 Dec '19

17 Dec '19

Hi everyone, Have anyone experienced kamcmd getting stuck and the only way to fix it was to restart the process? Running 5.2.5 Thanks -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 2

OPTION pings with dispatcher to asterisk box
by Karsten Horsmann 16 Dec '19

16 Dec '19

Hello all! i am using kamailio 5.3.1-4 on CentOS and i am using dispatcher to ping his targets. Now i have someone with asterisk that answers with 401 or 404 (depends on asterisk version). So i tried code=401;code=404 as aditional attrs. Seems not to work. Also i tired to change the modparam from class=2 to this: modparam("dispatcher", "ds_ping_reply_codes", "class=2,code=401,code=404") Is that a "bug" or a hidden feature, or i missed something? Btw - Have a nice weekend! SELECT * FROM dispatcher WHERE id=1 \G *************************** 1. row *************************** id: 1 setid: 1100 destination: sip:SOMEWHERE;transport=tls flags: 0 priority: 0 attrs: socket=tls:MY.IP.AD.RR:5061;ping_from=sip:MY.SIP.DOMAIN;code=401;code=404 description: UNTRUSTED "SET": { "ID": 1100, "TARGETS": [{ "DEST": { "URI": "sip:SOMEWHERE;transport=tls", "FLAGS": "IP", "PRIORITY": 0, "ATTRS": { "BODY": "socket=tls:MY.IP.AD.RR:5061;ping_from=sip:MY.SIP.DOMAIN;code=401;code=404", "DUID": "", "MAXLOAD": 0, "WEIGHT": 0, "RWEIGHT": 0, "SOCKET": "tls:MY.SIP.DOMAIN:5061" }, "LATENCY": { "AVG": 48768.816406, "STD": 139490.953125, "EST": 43359.566406, "MAX": 568876, "TIMEOUT": 8 } } }] } }] #!ifdef WITH_MYSQL modparam("dispatcher", "db_url", "DBURL") modparam("dispatcher", "table_name", "dispatcher") #!endif #!ifndef WITH_MYSQL modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") #!endif modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", "<sip:ping@FANCY-COMPANY-WEBNAME>") modparam("dispatcher", "ds_ping_interval", 30) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshold", 2) modparam("dispatcher", "setid_pvname", "$var(setid)") modparam("dispatcher", "attrs_pvname", "$var(attrs)") modparam("dispatcher", "ds_ping_reply_codes", "class=2,code=401,code=404") modparam("dispatcher", "ds_ping_latency_stats", 1) https://community.asterisk.org/t/asterisk-responds-with-404-to-options-even… Dec 13 17:25:46 siptrunk3 /usr/sbin/kamailio[25115]: INFO: <script>: Sending message 'OPTIONS sip:SOMEWHERE;transport=tls SIP/2.0 Via: SIP/2.0/TLS MY.IP.AD.RR:5061;branch=z9hG4bK2527.825671c0000000000000000000000000.0 To: <sip:SOMEWHERE;transport=tls> From: <sip:MY.SIP.DOMAIN>;tag=f9bf2579cb456ea8d260d4aabc8248ef-9ca844ad CSeq: 10 OPTIONS Call-ID: 01cb6e272da7b0dd-25115(a)MY.IP.AD.RR Max-Forwards: 70 Content-Length: 0 User-Agent: SBC-OS ' Dec 13 17:25:46 siptrunk3 /usr/sbin/kamailio[25127]: INFO: <script>: Received message 'SIP/2.0 404 Not Found Via: SIP/2.0/TLS MY.IP.AD.RR:5061;branch=z9hG4bK2527.825671c0000000000000000000000000.0;received=MY.IP.AD.RR From: <sip:MY.SIP.DOMAIN>;tag=f9bf2579cb456ea8d260d4aabc8248ef-9ca844ad To: <sip:SOMEWHERE;transport=tls>;tag=as54adca41 Call-ID: 01cb6e272da7b0dd-25115(a)MY.IP.AD.RR CSeq: 10 OPTIONS Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 ' -- Cheers *Karsten Horsmann*

3 6

Reject tls invites without a=crypto sdp
by Karsten Horsmann 16 Dec '19

16 Dec '19

Hi, I use Kamailio 5.3.1 with rtpengine to offer an siptrunk endpoint for my customers. I observe that someone of them use tls to encrypt signaling but forgotten to encrypt rtp. I want to reject this invites. Are there any hints how to do this? Thought about reading the sdp and search for a=crypto line and if not send reply with (what code ever will be good for that). Cheers Karsten

2 4

Removing extra Connection information in SDP
by Kjeld Flarup 16 Dec '19

16 Dec '19

Hello I have a problem with some phone numbers at Telenor. Our door telephone does not reply with an ACK on the 200 OK. All I found was a weird SDP response with two "connection information" lines Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): BroadWorks 307119541 1 IN IP4 194.247.61.31 Session Name (s): - Connection Information (c): IN IP4 194.247.61.31 Time Description, active time (t): 0 0 Session Attribute (a): sendrecv Media Description, name and address (m): audio 23116 RTP/AVP 8 101 Connection Information (c): IN IP4 194.247.61.31 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 Media Attribute (a): maxptime:40 Media Attribute (a): bsoft:1 image udptl t38 Media Attribute (a): sendrecv Media Attribute (a): silenceSupp:off - - - - Media Attribute (a): setup:actpass To debug, I would like to remove one of the "Connection Information (c): IN IP4 194.247.61.31", to see if this is what confuses the door telephone. But how do I do that? I tried with this but nothing happeded onreply_route[DOOR_MANAGE_REPLY] { if (is_method("INVITE|SUBSCRIBE")) { sdp_remove_line_by_prefix("c=IN IP4 194"); } return 1; } -- --------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen -www.liberalismen.dk <http://www.liberalismen.dk>

2 1

using tls targets in dispatcher
by Narasimhaiah, Vinod Maralenahalli 15 Dec '19

15 Dec '19

Hi I am using kamailio server to forward the request to a new kamailio-server which listens only on TLS. *problem: Not able to specify the dispatcher with tls protocol* I am using the below configuration in dispatcher.list kamailio version:4.4.2 *# setid(integer) destination(sip uri) flags (integer, optional), priority(int,opt), attrs (str,optional)1 sip:mynetwork.com <http://mynetwork.com>;transport=tls 8 1 socket=tls:mynetwork.com:5328 <http://mynetwork.com:5328>* and if I specify the destination as just sip:mynetwork.com:5328 with out attrs and transport=tls if is falling back to UDP. Thank you vinod.M.N

2 1

Kamailio dispatcher to use tls targets.
by vinod mn 15 Dec '19

15 Dec '19

Hi I am using kamailio server to forward the request to a new kamailio-server which listens only on TLS. *problem: Not able to specify the dispatcher with tls protocol* I am using the below configuration in dispatcher.list kamailio version:4.4.2 *# setid(integer) destination(sip uri) flags (integer, optional), priority(int,opt), attrs (str,optional)1 sip:mynetwork.com <http://mynetwork.com/>;transport=tls 8 1 socket=tls:mynetwork.com:5328 <http://mynetwork.com:5328/>* and if I specify the destination as just sip:mynetwork.com:5328 with out attrs and transport=tls if is falling back to UDP. Thank you vinod.M.N

1 0

using a pseudo-variable as a parameter to a module function
by Michael Iedema 14 Dec '19

14 Dec '19

Hi all, I’m trying to write a basic Kamailio module from scratch and have run into an area where I’m in need of a couple pointers (pun not intended). :-) I have the module loading correctly and can call a function in my routing script like this: mymod_getinfo(“5551234”); This, for now, just triggers a print of the submitted parameter. What I’d like to do next is have it accept and print the From URI in configuration script like so: mymod_getinfo(“$fU”); For now, this prints the string literal of “$fU” and I’m trying to extract the value of the referenced variable and print that. The example module “print” does not involve this concept of pseudo-variables. The skeleton module example in the wiki also does not mention variables. I’ve looked into other modules to see how it works but they all seem to have a slightly different handling of this task. Can someone point me to an example in the modules which accomplishes this? What I’ve been searching for is a replace_all function which accepts a string containing pseudo-variables and returns a new string with them all replaced with their corresponding values. Thanks in advance and apologies if I’ve overlooked a relevant piece of developer documentation or if I’m looking at this incorrectly. Regards, -Michael

3 3

lookup(aliases) issues with 5.2
by igor.potjevlesch＠gmail.com 13 Dec '19

13 Dec '19

Hello! I have notice some issues with lookup(aliases) since I upgraded to 5.2 from 5.1: First with kamctl: If this default command is executed: ctl_cmd_run ul.add "$USRLOC_TABLE" "$OSERUSER@$OSERDOMAIN" "$2" \ "$UL_EXPIRES" "$DEFAULT_Q" "$UL_PATH" "$UL_FLAGS" "$BR_FLAGS" "$ALL_METHODS" as $ALL_METHODS is not defined, there is a huge value in methods and then lookup(aliases) fails with -2. I have to update kamctl script and put -1 ("-1") instead of the variable. So, now, methods is a NULL value in DB and my alias is not ignored. But I use to make my alias like this: sip:number@blabla_1.local Which is no longer works: ERROR: registrar [common.c:62]: extract_aor(): failed to parse AoR [sip:number@blabla_1.local] Any idea of the difference since 5.2 with this behaviour? Without the underscore, the AoR seems to be valid but I need to support underscore as before. Regards, Igor.

4 16

t_newtrans() README issue
by Juha Heinanen 12 Dec '19

12 Dec '19

I noticed this in t_newtrans() README: Note that any flag operations (e.g. for accounting) after this function has been called will be ignored. You can use the the tmx module function t_flush_flags() to flush the altered flags to the already created transaction. The text is wrong. I have for ages set bflags after calliing t_newtrans() and the sets are NOT ignored. Here is an example: In route blocks: Dec 12 08:41:53 /usr/bin/sip-proxy[231131]: INFO: ********* called t_newtrans() Dec 12 08:41:53 /usr/bin/sip-proxy[231131]: INFO: ************ set bflag 9 In failure route block: Dec 12 08:41:58 /usr/bin/sip-proxy[231195]: INFO: ************ bflag 9 is set -- Juha

3 2

S-CSCF tear down call while deleting contact
by Liu, Charles 12 Dec '19

12 Dec '19

Hi All, I need some expert help to trouble shoot a problem I have seen, S-CSCF send BYE to tear down a call. I have make some progress to trace the problem to impurecord.c in ims_usrloc_scscf module, _c->first_dialog_data !=0 while try to delete an expired contact. I need some help to find out why _c->first_dialog_data is not 0 for an expired contact. To reproduce the problem, I began with the following for user1. Contact id 4098, 4114, and 4115 would expire and a call was using id 4116. When the time to delete contact, somehow scscf mistakenly think id 4098 still has a dialog, but the dialog was with id 4116 which was NOT expired. Therefore, S-CSCF send BYE to tear down the call, so end user see call drop. Please note, if id 4098 never make calls before, first call was working fine. But once a call was make, you will see this problem for next call. it seem S-CSCF did not clean up the dialog_data from last call properly and some data still hanging with the contact caused this problem. any insight how to fix or avoid this would be appreciated. id contact params path received user_agent expires callid 2192 sip:user2@192.168.164.136:62837;rinstance=bca96a7c77f53916;transport=UDP NULL <sip:term@pcscf.ims.mnc001.mcc001.3gppnetwork.org;lr> Zoiper rv2.10.3.2 2019-12-12 15:50:10 UPIM_gl22a13wlo47aeHRQ.. 4098 sip:user1@192.168.164.9:61946;rinstance=7ab86c5e85643767;transport=UDP NULL <sip:term@pcscf.ims.mnc001.mcc001.3gppnetwork.org;lr> Zoiper rv2.10.3.2 2019-12-12 15:49:21 ocl7UYxJLo8TnMw2p2-6NQ.. 4114 sip:user1@192.168.164.9:61946;rinstance=65afabdb886de0a7;transport=UDP NULL <sip:term@pcscf.ims.mnc001.mcc001.3gppnetwork.org;lr> Zoiper rv2.10.3.2 2019-12-12 15:50:11 GZrZHDyL3ojhDlLDPpJnvQ.. 4115 sip:user1@192.168.164.9:61946;rinstance=074e2e926212237b;transport=UDP NULL <sip:term@pcscf.ims.mnc001.mcc001.3gppnetwork.org;lr> Zoiper rv2.10.3.2 2019-12-12 15:50:13 DCKsqvUjZ1dAfJLfJ3nOhA.. 4116 sip:user1@192.168.164.9:61946;rinstance=38cc0c6188509aea;transport=UDP NULL <sip:term@pcscf.ims.mnc001.mcc001.3gppnetwork.org;lr> Zoiper rv2.10.3.2 2019-12-12 15:50:15 gF3fphqyrSbQwaN0eW4JLw.. 13(18) DEBUG: ims_usrloc_scscf [udomain.c:384]: mem_timer_udomain(): *** mem_timer_udomain - checking IMPUs - FINISHED *** 13(18) DEBUG: ims_usrloc_scscf [udomain.c:406]: mem_timer_udomain(): deleting contact [sip:user1@ims.mnc001.mcc001.3gppnetwork.org] 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:598]: delete_scontact(): Deleting contact: [sip:user1@192.168.164.9:61946;rinstance=074e2e926212237b;transport=UDP] 13(18) DEBUG: ims_usrloc_scscf [usrloc_db.c:370]: db_delete_ucontact(): Deleting ucontact [sip:user1@192.168.164.9:61946;rinstance=074e2e926212237b;transport=UDP] 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:310]: mem_delete_ucontact(): Checking if dialog_data is there and needs to be torn down 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:312]: mem_delete_ucontact(): first dialog is 0! 13(18) DEBUG: ims_usrloc_scscf [udomain.c:406]: mem_timer_udomain(): deleting contact [sip:user1@ims.mnc001.mcc001.3gppnetwork.org] 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:598]: delete_scontact(): Deleting contact: [sip:user1@192.168.164.9:61946;rinstance=7ab86c5e85643767;transport=UDP] 13(18) DEBUG: ims_usrloc_scscf [usrloc_db.c:370]: db_delete_ucontact(): Deleting ucontact [sip:user1@192.168.164.9:61946;rinstance=7ab86c5e85643767;transport=UDP] 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:310]: mem_delete_ucontact(): Checking if dialog_data is there and needs to be torn down 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:314]: mem_delete_ucontact(): first dialog is not 0 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:317]: mem_delete_ucontact(): Going to tear down dialog with info h_entry [1081] h_id [2579] 13(18) DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7f803f803108 with 1 -> 3 13(18) DEBUG: ims_dialog [dlg_hash.c:889]: lookup_dlg(): dialog id=2579 found on entry 1081 13(18) DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7f803f803108 with 1 -> 2 13(18) DEBUG: ims_dialog [dlg_req_within.c:379]: dlg_terminate(): terminating confirmed dialog Thanks, --Charles CONFIDENTIALITY NOTICE: This email and any attachments are for the sole use of the intended recipient and may contain material that is proprietary, confidential, privileged or otherwise legally protected or restricted under applicable government laws. Any review, disclosure, distributing or other use without expressed permission of the sender is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies without reading, printing, or saving.

1 0

Kamailio delayed reponse in case of database load
by Ali Taher 12 Dec '19

12 Dec '19

Hello, I have Kamailio installed as SIP redirect for an SBC to make routing decisions. I'm using SQLOps module with postgresql 11.5 where I have multiple tables containing each around 6M record of routing codes: Code options 392342 sup1|sup2|sup3 Where code field is of prefix_range data type and has a gist index. I'm sending traffic of 300 CPS to SBC which is forwarding it to Kamailio, and Kamailio respond with 300 multiple choice with the routing decision. In normal cases, everything is going smoothly where Kamailio repsonds up to 50 ms. But during the process of loading data to the routing tables, Kamailio response to SBC is dramatically delayed , where for some calls it takes up to 7 seconds to respond. Noting that during this process the data are loaded to staging tables and not to live tables used by Kamailio. Kamailio is installed on a VM with 32GB RAM and 16 vcores. Following configuration is used in Kamailio: fork=yes children=10 following are the customized parameters in postgresql.conf max_connections = 100 shared_buffers = 8GB effective_cache_size = 24GB maintenance_work_mem = 2GB checkpoint_completion_target = 0.7 wal_buffers = 16MB default_statistics_target = 100 random_page_cost = 1.1 effective_io_concurrency = 200 work_mem = 5242kB min_wal_size = 1GB max_wal_size = 2GB max_worker_processes = 16 max_parallel_workers_per_gather = 8 max_parallel_workers = 16 Following is the routing logic I'm using: route{ if (is_method("INVITE")) { $var(inc) = $(ct{param.value,tgrp}); xlog("L_INFO","CONTACT : $var(inc)" ); xlog("L_INFO","from number : $rU" ); if($var(inc)!=0) route(customer); route(block); } } route[customer] { sql_query("cb", "select caid from customeridentification where trunk='$var(inc)'", "ra"); $var(cid)=$dbr(ra=>[0,0]); sql_result_free("ra"); xlog("L_INFO","cid: $var(cid) \n"); sql_query("cb", "select routetablename from carrieraccount where caid=$var(cid)", "ra"); $var(tid)=$dbr(ra=>[0,0]); sql_result_free("ra"); if($var(cid)!=0) route(query); else route(block); } route[query] { #sql_query("cb", "select supplierid from route_$var(cid) where prefix @>'$rU' order by prefix limit 1", "ra"); sql_query("cb", "select get_routing('$var(tid)','$rU')", "ra"); $var(sup)=$dbr(ra=>[0,0]); sql_result_free("ra"); xlog("L_INFO","supplieris: $var(sup) \n"); #$var(x)=$(var(sup){s.replace,|,}{s.len})/4; $var(a)=$(var(sup){s.len}); $var(z)=$(var(sup){s.replace,|,}); $var(b)=$(var(z){s.len}); $var(suplen)=$var(a) - $var(b); if($var(sup)!=0) route(send); else route(block); } route[send] { xlog("L_INFO","Inside Send \n"); $var(i)=0; append_to_reply("Contact:"); while($var(i)<$var(suplen) + 1) { $var(s)=$(var(sup){s.select,$var(i),|}); $var(s)=$(var(s){s.select,0,;}); append_to_reply("<sip:$rU;tgrp=$(var(s){s.ltrim})@ xxx.xxx.xxx.xxx:5060;user=phone>"); if($var(i)<$var(suplen)) append_to_reply(","); $var(i)=$var(i)+1; } append_to_reply("\r\n"); sl_send_reply("300","Multiple Choices"); exit; } route[block] { xlog("L_INFO","INSIDE BLOCK" ); append_to_reply("Contact:<sip:$rU;tgrp=999999@xxx.xxx.xxx.xxx:5060;user=phone>\r\n"); sl_send_reply("300","Multiple Choices"); exit; } How can we enhance the Kamailio response in case of load on the postgresql server. Regards, Ali Taher

2 4

xhttp_prom Metric Naming Issue
by ENGIN ESIN 12 Dec '19

12 Dec '19

Hi everyone, I am trying to use xhttp_prom and there are some issues that i want to share with you. Metric Naming: I get warning at prometheus side like as follows, WARN[0659] append failed err="no token found" source="scrape.go:648" target="{__address__="192.168.42.1:9080", __metrics_path__="/metrics", __scheme__="http", instance="192.168.42.1:9080", job="vmm"}" With respect to prometheus documentation, metric names need to fullfill following naming requirement -> "It must match the regex [a-zA-Z_:][a-zA-Z0-9_:]*" (source: https://prometheus.io/docs/concepts/data_model/#metric-names-and-labels) However, kamailio gives unmatch outputs as follows because they have '-' char, (related discussion: https://github.com/prometheus/prometheus/issues/3154 ) * kamailio_usrloc_location-contacts 0 1576046310669 * kamailio_usrloc_location-expires 0 1576046310669 * kamailio_usrloc_location-users 0 1576046310669 whit that naming, prometheus can not parse values of any metrics. Documentation: Example 1.1. Set xhttp_prom_buf_size parameter ... modparam("xhttp", "xhttp_prom_buf_size", 1024) ... For that example, it should be as follows, ... modparam("xhttp_prom", "xhttp_prom_buf_size", 1024) ... [http://www.turkcell.com.tr/downloads/bireysel/img/Tcelldis.gif] <http://turkcell.li/iyaani> Bu elektronik posta ve onunla iletilen butun dosyalar sadece gondericisi tarafindan almasi amaclanan yetkili gercek ya da tuzel kisinin kullanimi icindir. Eger soz konusu yetkili alici degilseniz bu elektronik postanin icerigini aciklamaniz, kopyalamaniz, yonlendirmeniz ve kullanmaniz kesinlikle yasaktir ve bu elektronik postayi derhal silmeniz gerekmektedir. TURKCELL bu mesajin icerdigi bilgilerin do?rulu?u veya eksiksiz oldugu konusunda herhangi bir garanti vermemektedir. Bu nedenle bu bilgilerin ne sekilde olursa olsun iceriginden, iletilmesinden, alinmasindan ve saklanmasindan sorumlu degildir. Bu mesajdaki gorusler yalnizca gonderen kisiye aittir ve TURKCELLin goruslerini yansitmayabilir Bu e-posta bilinen butun bilgisayar viruslerine karsi taranmistir. ________________________________ This e-mail and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you are not the intended recipient you are hereby notified that any dissemination, forwarding, copying or use of any of the information is strictly prohibited, and the e-mail should immediately be deleted. TURKCELL makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the information transmission, reception, storage or use of such in any way whatsoever. The opinions expressed in this message belong to sender alone and may not necessarily reflect the opinions of TURKCELL. This e-mail has been scanned for all known computer viruses.

2 1

Please Help
by Sharath 12 Dec '19

12 Dec '19

Dear Team, If we want add a free-pbx as a trunk then what are configuration we need to do in admin panel of siremis. * SHARATH. O.M.** *

2 1

Kamailio delayed reponse in case of database load
by Ali Taher 10 Dec '19

10 Dec '19

Hello, I have Kamailio installed as SIP redirect for an SBC to make routing decisions. I'm using SQLOps module with postgresql 11.5 where I have multiple tables containing each around 6M record of routing codes: Code options ----------------------- 392342 sup1|sup2|sup3 Where code field is of prefix_range data type and has a gist index. I'm sending traffic of 300 CPS to SBC which is forwarding it to Kamailio, and Kamailio respond with 300 multiple choice with the routing decision. In normal cases, everything is going smoothly where Kamailio repsonds up to 50 ms. But during the process of loading data to the routing tables, Kamailio response to SBC is dramatically delayed , where for some calls it takes up to 7 seconds to respond. Noting that during this process the data are loaded to staging tables and not to live tables used by Kamailio. Kamailio is installed on a VM with 32GB RAM and 16 vcores. Following configuration is used in Kamailio: fork=yes children=10 following are the customized parameters in postgresql.conf max_connections = 100 shared_buffers = 8GB effective_cache_size = 24GB maintenance_work_mem = 2GB checkpoint_completion_target = 0.7 wal_buffers = 16MB default_statistics_target = 100 random_page_cost = 1.1 effective_io_concurrency = 200 work_mem = 5242kB min_wal_size = 1GB max_wal_size = 2GB max_worker_processes = 16 max_parallel_workers_per_gather = 8 max_parallel_workers = 16

4 6

authcheck and fd variable
by Robert Lieback 10 Dec '19

10 Dec '19

Hello, i'm looking into kamalio (btw: thanks to asipto for the great book) and i hope someone can push me in the right direction. I have a question/problem with some lines in the default config (kamailio.cfg) around line 754: ... if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } ... auth_check() and auth_challenge doesn't work, but i've provided the correct credentials. When i replace "$fd" with domain "voip.XXXXXX.de" it works. This domain is also defined as alias in the config and all sip messages i send to the server have a correct from header with this domain. First step for me would be to look whats stored in $fd, but how can i output the content of $fd? Cause log(2, "$fd") or xlog("$fd") doesn't work and throws errors about unknown variables. Thanks, Robert Lieback

2 2

Kamailio management group updates
by Daniel-Constantin Mierla 09 Dec '19

09 Dec '19

Hello, Jesus Rodrigues and Inaki Baz Castillo changed their work roles a while ago, so they no longer deal with Kamailio and they do not expect to do it in the near future. To keep the Management group with members connected as much as possible with the project, Fred Posner and Victor Seva were proposed to replace them. For the new members in the public lists: the management group of Kamailio is not involved in the daily development or administration of the project -- all those are done on public mailing lists or kadmin group (internal infrastructure administration) -- management team is supposed to take the decisions on sensitive situations for the evolution of the project. Should anyone have opinions about these changes, let's discuss on sr-users mailing list. 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

1 0

duplicated registrations
by Liu, Charles 08 Dec '19

08 Dec '19

Hi All, Does this look correct? I am using Zoiper client and did multiple registration/re-registration, but I saw duplicated contacts in the database for the same user/device(alice). I was expecting to see only one. Is this a bug? How do I prevent this happening? "select * from scscf.contact" id contact params path received user_agent expires callid 49 sip:bob@192.168.10.201:60944;rinstance=44b4024df17bd284;transport=UDP NULL <sip:term@pcscf.mnc001.mnc001.3gppnetwork.org;lr> Zoiper rv2.10.3.2 2019-12-08 10:31:51 r7jCYp7JuVbA8QjocnouLw.. 64 sip:alice@192.168.10.241:60383;rinstance=e850d3ed0605bcd3;transport=UDP NULL <sip:term@pcscf.mnc001.mnc001.3gppnetwork.org;lr> Zoiper rv2.10.3.2 2019-12-08 10:31:11 EHLOADkLUEe8d0CMpdLL9A.. 142 sip:alice@192.168.10.241:60383;rinstance=bc755d87cc438890;transport=UDP NULL <sip:term@pcscf.mnc001.mnc001.3gppnetwork.org;lr> Zoiper rv2.10.3.2 2019-12-08 10:31:51 P1rBkHGUVdv7oqW3_UWQPg.. 143 sip:alice@192.168.10.241:60383;rinstance=146303cf7501a125;transport=UDP NULL <sip:term@pcscf.mnc001.mnc001.3gppnetwork.org;lr> Zoiper rv2.10.3.2 2019-12-08 10:32:15 etUEoAGSM4q4e_39wFKa7w.. 144 sip:alice@192.168.10.241:60383;rinstance=e0a260b7b78ecf6b;transport=UDP NULL <sip:term@pcscf.mnc001.mnc001.3gppnetwork.org;lr> Zoiper rv2.10.3.2 2019-12-08 10:32:17 WUtIYeGlBZmHD3TGklmKCQ.. 145 sip:alice@192.168.10.241:60383;rinstance=2681245431c4e4e4;transport=UDP NULL <sip:term@pcscf.mnc001.mnc001.3gppnetwork.org;lr> Zoiper rv2.10.3.2 2019-12-08 10:32:19 HNZPyeO173jfscKYNOT0bw.. Thanks, --Charles CONFIDENTIALITY NOTICE: This email and any attachments are for the sole use of the intended recipient and may contain material that is proprietary, confidential, privileged or otherwise legally protected or restricted under applicable government laws. Any review, disclosure, distributing or other use without expressed permission of the sender is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies without reading, printing, or saving.

2 2

ul.dump empty
by Liu, Charles 07 Dec '19

07 Dec '19

Hi All, I am running Kamailio v5.2.4 with P/S/I-cscf instances on 3 different machines with default IMS example configurations. Kamcmd ul.dump show empty, but I can make calls... any idea what is the problem? root@pcscf:/# kamcmd ulpcscf.status { Domain: location Size: 512 AoRs: { Stats: { Records: 7 Max-Slots: 6 } } } root@pcscf:/# kamcmd ul.dump { Domains: { } } CONFIDENTIALITY NOTICE: This email and any attachments are for the sole use of the intended recipient and may contain material that is proprietary, confidential, privileged or otherwise legally protected or restricted under applicable government laws. Any review, disclosure, distributing or other use without expressed permission of the sender is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies without reading, printing, or saving.

1 1

Pike module never release IP
by Andrew Chen 06 Dec '19

06 Dec '19

Hi all, Have anyone ever experienced pike module not releasing the IP after "remove_latency" period? We ran a load test and this is what happened to one of the two paired nodes and I had to restart kamailio process to release it. I'm running Kam 5.2.5 Thanks. -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 6

How can I make calls between the two domains?
by Yanko 06 Dec '19

06 Dec '19

Hello: I have two virtual machines with two kamailo domains VM1: Domain: 172.16.16.1 Users:2000, 2001, 2002... VM2: Domain: 172.16.16.2 Users:3000, 3001, 3002... In each domain I can make calls. For example, user 2000 can call 2001. User 3000 can call 3001. How can I make calls between the two domains? What modules do I need to load? How to configure the kamailio.cfg file? Thanks, YANKO.

2 1

Scripts still valid?
by David Villasmil 05 Dec '19

05 Dec '19

Hello guys, Is this still valid? https://github.com/kamailio/kamailio/blob/master/misc/examples/scripts/ctd.… Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 4

call disconnected between 15 to 45 minutes
by Liu, Charles 05 Dec '19

05 Dec '19

Hi all, I am still looking for some help to debug why S-CSCF send BYE to caller and callee after a call was connected. from wireshark, I just saw S-CSCF send BYE to both caller and callee. That matched the S-CSCF debug output. But the question is why S-CSCF delete contact and send BYE to a active call? There was no timer expire or something like that in the log... Anyone can help? Thanks, --Charles 19(24) DEBUG: <core> [core/mem/q_malloc.c:519]: qm_free(): freeing frag. 0x7fb09413e6d0 alloc'ed from cdp: diameter_msg.c: AAATranslateMessage(484) 19(24) DEBUG: tm [t_lookup.c:1618]: t_lookup_ident(): transaction found 19(24) DEBUG: <core> [core/mem/q_malloc.c:367]: qm_malloc(): qm_malloc(0x7fb093db6000, 6872) called from core: core/sip_msg_clone.c: sip_msg_shm_clone(496) 19(24) DEBUG: <core> [core/mem/q_malloc.c:412]: qm_malloc(): qm_malloc(0x7fb093db6000, 6872) returns address 0x7fb09414d450 frag. 0x7fb09414d418 (size=6872) on 1 -th hit 19(24) ERROR: *** cfgtrace:failure_route=[REG_SAR_REPLY] c=[/etc/kamailio/kamailio.cfg] l=767 a=26 n=xlog 19(24) DEBUG: <script>: saa_return code is 1 19(24) ERROR: *** cfgtrace:failure_route=[REG_SAR_REPLY] c=[/etc/kamailio/kamailio.cfg] l=769 a=26 n=xlog 19(24) DEBUG: <script>: saa_return code (for scscf_save on register) is 1 19(24) ERROR: *** cfgtrace:failure_route=[REG_SAR_REPLY] c=[/etc/kamailio/kamailio.cfg] l=783 a=21 n=switch 19(24) ERROR: *** cfgtrace:failure_route=[REG_SAR_REPLY] c=[/etc/kamailio/kamailio.cfg] l=772 a=26 n=xlog 19(24) DEBUG: <script>: SAR success - 200 response sent from module 19(24) ERROR: *** cfgtrace:failure_route=[REG_SAR_REPLY] c=[/etc/kamailio/kamailio.cfg] l=773 a=2 n=exit 19(24) DEBUG: <core> [core/mem/q_malloc.c:475]: qm_free(): qm_free(0x7fb093db6000, 0x7fb09414d450), called from tm: t_reply.c: free_faked_req(948) 19(24) DEBUG: <core> [core/mem/q_malloc.c:519]: qm_free(): freeing frag. 0x7fb09414d418 alloc'ed from core: core/sip_msg_clone.c: sip_msg_shm_clone(496) 19(24) DEBUG: <core> [core/mem/q_malloc.c:475]: qm_free(): qm_free(0x7fb093db6000, 0x7fb09413e978), called from ims_registrar_scscf: cxdx_sar.c: free_saved_transaction_data(94) 19(24) DEBUG: <core> [core/mem/q_malloc.c:519]: qm_free(): freeing frag. 0x7fb09413e940 alloc'ed from ims_registrar_scscf: save.c: save(1414) 19(24) DEBUG: <core> [core/mem/q_malloc.c:475]: qm_free(): qm_free(0x7fb093db6000, 0x7fb09413e0b0), called from ims_registrar_scscf: reply.c: free_contact_buf(830) 19(24) DEBUG: <core> [core/mem/q_malloc.c:519]: qm_free(): freeing frag. 0x7fb09413e078 alloc'ed from ims_registrar_scscf: reply.c: build_contact(443) 19(24) DEBUG: <core> [core/mem/q_malloc.c:475]: qm_free(): qm_free(0x7fb093db6000, 0x7fb094145218), called from ims_registrar_scscf: reply.c: free_contact_buf(835) 19(24) DEBUG: <core> [core/mem/q_malloc.c:519]: qm_free(): freeing frag. 0x7fb0941451e0 alloc'ed from ims_registrar_scscf: reply.c: build_contact(435) 19(24) DEBUG: <core> [core/mem/q_malloc.c:475]: qm_free(): qm_free(0x7fb093db6000, 0x7fb094147dc0), called from ims_registrar_scscf: cxdx_sar.c: free_saved_transaction_data(98) 19(24) DEBUG: <core> [core/mem/q_malloc.c:519]: qm_free(): freeing frag. 0x7fb094147d88 alloc'ed from ims_registrar_scscf: save.c: save(1396) 19(24) DEBUG: <core> [core/mem/q_malloc.c:475]: qm_free(): qm_free(0x7fb093db6000, 0x7fb09415b4c8), called from cdp: transaction.c: cdp_free_trans(182) 19(24) DEBUG: <core> [core/mem/q_malloc.c:519]: qm_free(): freeing frag. 0x7fb09415b490 alloc'ed from cdp: transaction.c: cdp_add_trans(110) 19(24) DEBUG: <core> [core/mem/q_malloc.c:475]: qm_free(): qm_free(0x7fb093db6000, 0x7fb0941442a0), called from cdp: transaction.c: cdp_free_trans(183) 19(24) DEBUG: <core> [core/mem/q_malloc.c:519]: qm_free(): freeing frag. 0x7fb094144268 alloc'ed from cdp: transaction.c: cdp_add_trans(105) 13(18) DEBUG: ims_usrloc_scscf [ims_usrloc_scscf_mod.c:481]: timer(): Syncing cache 13(18) DEBUG: ims_usrloc_scscf [udomain.c:291]: mem_timer_udomain(): *** mem_timer_udomain - checking contacts - START *** 13(18) DEBUG: ims_usrloc_scscf [udomain.c:301]: mem_timer_udomain(): We have a [3gpp=0] contact in the new contact list in slot 95 = [sip:bob@ims.mnc001.mnc001.3gppnetwork.org] (sip:bob@192.168.164.248:50448;rinstance=989f943bd5098a11;transport=UDP) which expires in -60.000000 seconds and has a ref count of 0 (state: Contact with delayed delete) 13(18) DEBUG: ims_usrloc_scscf [udomain.c:309]: mem_timer_udomain(): contact in state CONTACT_DELATED_DELETE is about to be deleted13(18) DEBUG: ims_usrloc_scscf [udomain.c:301]: mem_timer_udomain(): We have a [3gpp=0] contact in the new contact list in slot 464 = [sip:alice@ims.mnc001.mnc001.3gppnetwork.org] (sip:alice@192.168.164.203:44640;rinstance=e35e9704884c3208;transport=UDP) which expires in 50.000000 seconds and has a ref count of 1 (state: Contact valid) 13(18) DEBUG: ims_usrloc_scscf [udomain.c:301]: mem_timer_udomain(): We have a [3gpp=0] contact in the new contact list in slot 478 = [sip:bob@ims.mnc001.mnc001.3gppnetwork.org] (sip:bob@192.168.164.248:50448;rinstance=9e742618a5f6bcae;transport=UDP) which expires in 56.000000 seconds and has a ref count of 1 (state: Contact valid) 13(18) DEBUG: ims_usrloc_scscf [udomain.c:353]: mem_timer_udomain(): *** mem_timer_udomain - checking contacts - FINISHED *** 13(18) DEBUG: ims_usrloc_scscf [udomain.c:359]: mem_timer_udomain(): *** mem_timer_udomain - checking IMPUs - START *** 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:372]: process_impurecord(): Checking validity of IMPU: <sip:alice@ims.mnc001.mnc001.3gppnetwork.org> registration subscriptions 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:404]: process_impurecord(): Public Identity sip:alice@ims.mnc001.mnc001.3gppnetwork.org, Barred: [0], State: [registered], contacts [1], 3gppcontacts [0] 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:441]: process_impurecord(): Contact #0 - sip:alice@192.168.164.203:44640;rinstance=e35e9704884c3208;transport=UDP, Ref [1] (expires in 50 seconds) (State: 0) 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:372]: process_impurecord(): Checking validity of IMPU: <sip:bob@ims.mnc001.mnc001.3gppnetwork.org> registration subscriptions 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:404]: process_impurecord(): Public Identity sip:bob@ims.mnc001.mnc001.3gppnetwork.org, Barred: [0], State: [registered], contacts [1], 3gppcontacts [0] 3(18) DEBUG: ims_usrloc_scscf [impurecord.c:441]: process_impurecord(): Contact #0 - sip:bob@192.168.164.248:50448;rinstance=9e742618a5f6bcae;transport=UDP, Ref [1] (expires in 56 seconds) (State: 0) 3(18) DEBUG: ims_usrloc_scscf [udomain.c:384]: mem_timer_udomain(): *** mem_timer_udomain - checking IMPUs - FINISHED *** 3(18) DEBUG: ims_usrloc_scscf [udomain.c:406]: mem_timer_udomain(): deleting contact [sip:bob@ims.mnc001.mnc001.3gppnetwork.org] 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:598]: delete_scontact(): Deleting contact: [sip:bob@192.168.164.248:50448;rinstance=989f943bd5098a11;transport=UDP] 13(18) DEBUG: ims_usrloc_scscf [usrloc_db.c:370]: db_delete_ucontact(): Deleting ucontact [sip:bob@192.168.164.248:50448;rinstance=989f943bd5098a11;transport=UDP] 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:310]: mem_delete_ucontact(): Checking if dialog_data is there and needs to be torn down 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:314]: mem_delete_ucontact(): first dialog is not 0 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:317]: mem_delete_ucontact(): Going to tear down dialog with info h_entry [1694] h_id [3889] 13(18) DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7fb09414ce40 with 1 -> 3 13(18) DEBUG: ims_dialog [dlg_hash.c:889]: lookup_dlg(): dialog id=3889 found on entry 1694 13(18) DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7fb09414ce40 with 1 -> 2 13(18) DEBUG: ims_dialog [dlg_req_within.c:379]: dlg_terminate(): terminating confirmed dialog 13(18) DEBUG: <core> [core/mem/q_malloc.c:367]: qm_malloc(): qm_malloc(0x7fb09b5ae010, 18) called from ims_dialog: dlg_req_within.c: build_extra_hdr(229) 13(18) DEBUG: <core> [core/mem/q_malloc.c:412]: qm_malloc(): qm_malloc(0x7fb09b5ae010, 24) returns address 0x7fb09b640c30 frag. 0x7fb09b640bf8 (size=64) on 1 -th hit 13(18) DEBUG: <core> [core/mem/q_malloc.c:367]: qm_malloc(): qm_malloc(0x7fb09b5ae010, 248) called from ims_dialog: dlg_req_within.c: build_dlg_t(81) 13(18) DEBUG: <core> [core/mem/q_malloc.c:412]: qm_malloc(): qm_malloc(0x7fb09b5ae010, 248) returns address 0x7fb09b74b100 frag. 0x7fb09b74b0c8 (size=248) on 1 -th hit 13(18) DEBUG: <core> [core/mem/q_malloc.c:367]: qm_malloc(): qm_malloc(0x7fb09b5ae010, 72) called from core: core/parser/parse_rr.c: do_parse_rr_body(68) 13(18) DEBUG: <core> [core/mem/q_malloc.c:412]: qm_malloc(): qm_malloc(0x7fb09b5ae010, 72) returns address 0x7fb09b74b048 frag. 0x7fb09b74b010 (size=80) on 1 -th hit 13(18) DEBUG: ims_dialog [dlg_req_within.c:272]: send_bye(): sending BYE to caller 13(18) DEBUG: ims_dialog [dlg_hash.c:1047]: ref_dlg(): ref dlg 0x7fb09414ce40 with 1 -> 3 CONFIDENTIALITY NOTICE: This email and any attachments are for the sole use of the intended recipient and may contain material that is proprietary, confidential, privileged or otherwise legally protected or restricted under applicable government laws. Any review, disclosure, distributing or other use without expressed permission of the sender is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies without reading, printing, or saving.

2 2

Question regarding priority in dispatcher module
by Floimair Florian 05 Dec '19

05 Dec '19

Dear all! After reading the documentation on dispatcher module I still have a question that isn't really documented. Regarding the priority value of an entry in a dispatcher list: Is a lower numerical priority value regarded as higher priority or is it the other way around? Hopefully someone can help me to clarify this. With best regards FLORIAN FLOIMAIR Software Development Commend International GmbH Saalachstrasse 51 5020 Salzburg, Austria commend.com LG Salzburg / FN 178618z

2 2

Re: [SR-Users] RTPengine with rejected RE-INVITE when port is changing
by Daniel Tryba 05 Dec '19

05 Dec '19

On Thu, Dec 05, 2019 at 09:37:51AM +0000, Laurent Schweizer wrote: > Hello, > > I already see old post about this : > https://opensips.org/pipermail/users/2014-November/030451.html > > but I???m interested to know if now they is a solution ???? > > so the issue is a RE-INVITE rejected (415 Unsupported Media or 488 ) and the RTP port is changing. > Is that case the old RTP relay (I???m using RTPengine) is destroyed after 30sec. > Any way to restore them ? I'm confused: - opensips != kamailio - rtpproxy != rtpengine But RFC 3261 states (page 16): During the session, either Alice or Bob may decide to change the characteristics of the media session. This is accomplished by sending a re-INVITE containing a new media description. This re- INVITE references the existing dialog so that the other party knows that it is to modify an existing session instead of establishing a new session. The other party sends a 200 (OK) to accept the change. The requestor responds to the 200 (OK) with an ACK. If the other party does not accept the change, he sends an error response such as 488 (Not Acceptable Here), which also receives an ACK. However, the failure of the re-INVITE does not cause the existing call to fail - the session continues using the previously negotiated characteristics. Full details on session modification are in Section 14. So if this happens with kamailio/rtpengine that is a bug in (I presume) rtpengine.

3 2

siputils: how does sip_p_charging_vector generate the header
by George Diamantopoulos 05 Dec '19

05 Dec '19

Hello, I've been looking at the sip_p_charging_vector() function of siputils. From the documentation it is not evident how the PCV header is generated. Are the exported $pcv pseudovariables writable, so that the function canl use those when generating the header? Thanks! BR, George

2 3

anycast + dmq_usrloc
by Nuno Miguel Reis 05 Dec '19

05 Dec '19

Hi guys. I'm setting up an anycast scenario with several kamailios sharing an active state and I've noticed that the dmq_usrloc module is not replication the usrloc socket_info attribute when a register comes in through one of the kamailio nodes. Is there a reason behind this? I'm thinking on adding the replication for that attribute in the code. Another way of address this woud be by having a modparam to replicate the socket. What do you think? This is what I'm seeing: *anycast node that actually gets the SIP REGISTER:* $ kamcmd ul.dump { Domains: { Domain: { Domain: location Size: 32768 AoRs: { Info: { AoR: 309(a)nmreis.test HashID: -35263718 Contacts: { Contact: { Address: sip:309@192.168.150.158:55060;transport=udp;registering_acc=nmreis_test Expires: 585 Q: -1.000000 Call-ID: 6e7635ce4604e16400ee763b2ead55b2@0:0:0:0:0:0:0:0 CSeq: 4 User-Agent: Jitsi2.11.5602Linux Received: sip: 2.2.2.2:31171 Path: [not set] State: CS_SYNC Flags: 1 CFlags: 192 Socket: udp: 1.1.1.1:5060 Methods: -1 Ruid: uloc-5de8056d-22ab-f Instance: [not set] Reg-Id: 0 Server-Id: 0 Tcpconn-Id: -1 Keepalive: 1 Last-Keepalive: 1575487540 Last-Modified: 1575487540 } } } } Stats: { Records: 1 Max-Slots: 1 } } } } *anycast node that gets the USRLOC update via KDMQ:* $ kamcmd ul.dump { Domains: { Domain: { Domain: location Size: 32768 AoRs: { Info: { AoR: 309(a)nmreis.test HashID: -35263718 Contacts: { Contact: { Address: sip:309@192.168.150.158:55060;transport=udp;registering_acc=nmreis_test Expires: 574 Q: -1.000000 Call-ID: 6e7635ce4604e16400ee763b2ead55b2@0:0:0:0:0:0:0:0 CSeq: 4 User-Agent: Jitsi2.11.5602Linux Received: sip: 2.2.2.2:31171 Path: [not set] State: CS_SYNC Flags: 1 CFlags: 192 Socket: [not set] Methods: -1 Ruid: uloc-5de7ff7b-4617-3 Instance: [not set] Reg-Id: 0 Server-Id: 0 Tcpconn-Id: -1 Keepalive: 1 Last-Keepalive: 1575487540 Last-Modified: 1575487540 } } } } Stats: { Records: 1 Max-Slots: 1 } } } } *the actual KDMQ message:* *KDMQ sip:usrloc@10.100.100.21:5060 <http://sip:usrloc@10.100.100.21:5060> SIP/2.0 Via: SIP/2.0/UDP 10.100.100.22;branch=z9hG4bK6034.290e4186000000000000000000000000.0 To: <sip:usrloc@10.100.100.21:5060 <http://sip:usrloc@10.100.100.21:5060>> From: <sip:usrloc@10.100.100.22:5060 <http://sip:usrloc@10.100.100.22:5060>>;tag=3393f0703fb0ccaca74109ff37de39f5-1e4c092b CSeq: 10 KDMQ Call-ID: 0cfde4d32a31addc-8875(a)127.0.0.1 <0cfde4d32a31addc-8875(a)127.0.0.1> Content-Length: 433 Max-Forwards: 1 Content-Type: application/json {"action":1,"aor":"309(a)nmreis.test","ruid":"uloc-5de8056d-22ab-f","c":"sip:309@192.168.150.158:55060;transport=udp;registering_acc=nmreis_test","received":"sip:2.2.2.2:31171 <http://2.2.2.2:31171>","path":"","callid":"6e7635ce4604e16400ee763b2ead55b2@0:0:0:0:0:0:0:0","user_agent":"Jitsi2.11.5602Linux","instance":"","expires":1575487600,"cseq":2,"flags":1,"cflags":192,"q":-1,"last_modified":1575487000,"methods":4294967295,"reg_id":0,"server_id":0}* -- Nuno Miguel Reis Departamento de Engenharia Informática Faculdade de Ciências e Tecnologia Universidade de Coimbra mobile: +351 918519145 -- get my PGP key @ http://pgp.dei.uc.pt:11371/pks/lookup?op=get&search=0x750296EECEAD2B6D

2 2

anycast + dmq_usrloc
by Nuno Miguel Reis 04 Dec '19

04 Dec '19

Hi guys. I'm setting up an anycast scenario with several kamailios sharing an active state and I've noticed that the dmq_usrloc module is not replicating the usrloc socket_info attribute when a register comes in through one of the kamailio nodes. Is there a reason behind this? I'm thinking on adding the replication for that attribute in the code. Another way of address this woud be by having a modparam to replicate the socket. What do you think? This is what I'm seeing: anycast node that actually gets the SIP REGISTER: $ kamcmd ul.dump { Domains: { Domain: { Domain: location Size: 32768 AoRs: { Info: { AoR: 309(a)nmreis.test HashID: -35263718 Contacts: { Contact: { Address: sip:309@192.168.150.158:55060;transport=udp;registering_acc=nmreis_test Expires: 585 Q: -1.000000 Call-ID: 6e7635ce4604e16400ee763b2ead55b2@0:0:0:0:0:0:0:0 CSeq: 4 User-Agent: Jitsi2.11.5602Linux Received: sip: 2.2.2.2:31171 [http://2.2.2.2:31171/] Path: [not set] State: CS_SYNC Flags: 1 CFlags: 192 Socket: udp: 1.1.1.1:5060 [http://1.1.1.1:5060/] Methods: -1 Ruid: uloc-5de8056d-22ab-f Instance: [not set] Reg-Id: 0 Server-Id: 0 Tcpconn-Id: -1 Keepalive: 1 Last-Keepalive: 1575487540 Last-Modified: 1575487540 } } } } Stats: { Records: 1 Max-Slots: 1 } } } } anycast node that gets the USRLOC update via KDMQ: $ kamcmd ul.dump { Domains: { Domain: { Domain: location Size: 32768 AoRs: { Info: { AoR: 309(a)nmreis.test HashID: -35263718 Contacts: { Contact: { Address: sip:309@192.168.150.158:55060;transport=udp;registering_acc=nmreis_test Expires: 574 Q: -1.000000 Call-ID: 6e7635ce4604e16400ee763b2ead55b2@0:0:0:0:0:0:0:0 CSeq: 4 User-Agent: Jitsi2.11.5602Linux Received: sip: 2.2.2.2:31171 [http://2.2.2.2:31171/] Path: [not set] State: CS_SYNC Flags: 1 CFlags: 192 Socket: [not set] Methods: -1 Ruid: uloc-5de7ff7b-4617-3 Instance: [not set] Reg-Id: 0 Server-Id: 0 Tcpconn-Id: -1 Keepalive: 1 Last-Keepalive: 1575487540 Last-Modified: 1575487540 } } } } Stats: { Records: 1 Max-Slots: 1 } } } } the actual KDMQ message: KDMQ sip:usrloc@10.100.100.21:5060 [http://sip:usrloc@10.100.100.21:5060/] SIP/2.0 Via: SIP/2.0/UDP 10.100.100.22;branch= z9hG4bK6034.290e4186000000000000000000000000.0 To: < sip:usrloc@10.100.100.21:5060 [http://sip:usrloc@10.100.100.21:5060/] > From: < sip:usrloc@10.100.100.22:5060 [http://sip:usrloc@10.100.100.22:5060/] >;tag= 3393f0703fb0ccaca74109ff37de39f5-1e4c092b CSeq: 10 KDMQ Call-ID: 0cfde4d32a31addc-8875(a)127.0.0.1 [mailto:0cfde4d32a31addc-8875@127.0.0.1] Content-Length: 433 Max-Forwards: 1 Content-Type: application/json {"action":1,"aor":"309(a)nmreis.test","ruid":"uloc-5de8056d-22ab-f","c":"sip:309@192.168.150.158:55060;transport=udp;registering_acc=nmreis_test","received":"sip: 2.2.2.2:31171 [http://2.2.2.2:31171/] ","path":"","callid":"6e7635ce4604e16400ee763b2ead55b2@0:0:0:0:0:0:0:0","user_agent":"Jitsi2.11.5602Linux","instance":"","expires":1575487600,"cseq":2,"flags":1,"cflags":192,"q":-1,"last_modified":1575487000,"methods":4294967295,"reg_id":0,"server_id":0} Looking forward to hear from you. BR, -- Nuno Miguel Reis Departamento de Engenharia Informática Faculdade de Ciências e Tecnologia Universidade de Coimbra

1 0

Vendor Specific RBT Call Flow
by Hamid Hashmi 04 Dec '19

04 Dec '19

Dear Users Can the following call flow be implemented, if yes, can you give some suggestions with modules information? [cid:d15f60e7-73ec-4cd8-8954-92f472a94ef7] Regards Hamid R. Hashmi

3 2

siremis help and usage index?
by PICCORO McKAY Lenz 04 Dec '19

04 Dec '19

it's there some help manual usage for siremis? the only KB source i have are asipto.. and only are install manuals!! Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

3 2

Re: [SR-Users] kamailio IMS supporting VOLTE
by Shubhendu Das 04 Dec '19

04 Dec '19

Hi , Does Kamailio has its own HSS supporting CX Interface ? BR, Shubhendu From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Mojtaba Sent: Saturday, November 23, 2019 4:37 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] kamailio IMS supporting VOLTE Hello, Running the IMS network for supporting VoLTE with Kamailio module and FHOSS is challenging. THe Kamailio works great in IMS network but You have to optimise your own database. Using FHOSS without any support or development is very risky. In general speaking, running VoLTE have three main section and the IMS network is one of them. You might have some challenges in EPC and RAN access. WIth Best Regards. Mojtaba On Thu, Nov 21, 2019 at 3:23 PM Shubhendu Das <shubhendu.das(a)azcom.in> wrote: Hi , I have a query regarding Kamailio open source SIP server. Does it have any release, where its supports the feature of IMS supporting VOLTE with real commercial mobile phones(IMSI based) ? If so, what is the release number ? Please also let me know about the supported mobile phones list. BR, Shubhendu Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- --Mojtaba Esfandiari.S -- _Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._

2 1

Fwd: app-ads.txt monetization proposal
by Joel Serrano 04 Dec '19

04 Dec '19

Not sure if this is BS or worth exploring, it's not my expertise area so I'm forwarding to you all just in case ---------- Forwarded message --------- From: Lorenzo from Instal <lorenzo.tacchio(a)instal.com> Date: Tue, Dec 3, 2019 at 8:56 AM Subject: app-ads.txt monetization proposal To: <joel(a)textplus.com> app-ads.txt monetization proposal Hello , I am Lorenzo, Senior Account Manager at Instal; we are currently buying your app's traffic via DSPs. We thought would be beneficial to work directly with you, *monetizing your inventory with no intermediate levels at all*. The key of our success is the *simple technical integration*. No SDK is required, we will add our ads.txt to your page and your app would be included in our media buying plan with budgets of *more than $1M per month*. Would you be interested to discover more in detail this opportunity? Please share your Skype ID or add me, *live:loresseintes*, to connect further! Kind Regards, Lorenzo Instal Srl, Via Ippolito Pindemonte, Florence, Italy Manage preferences <https://hs-3923849.s.hubspotemail.net/hs/manage-preferences/unsubscribe?d=V…>

1 1

secfilter usage
by Aymeric Moizard 03 Dec '19

03 Dec '19

Hi all, I'm testing and moving my kamailio script to use the newer secfilter module. In the past, I was replying "200 Ok" to "friendly scanners"... With the newer secfilter module, it looks like I can't send a reply with "sl_send_reply("200", "OK")" secf_check_ua(); if ($? == -2) { sl_send_reply("200", "OK"); exit; } I have read the code of the secfilter, but I was not able to see any specific code to silently discard the SIP request. I can see in the documentation about "secf_check_sqli_all", that the SIP message is supposed to be "dropped". I can see "w_check_sqli_all" returns 0 on detection and w_check_ua returns -2 upon detection. Are the message discarded because a negative value was returned? Would it be doable, using the secfilter, to still reply 200 Ok? Regards Aymeric -- Antisip - http://www.antisip.com

2 5

kamailio IMS supporting VOLTE
by Shubhendu Das 03 Dec '19

03 Dec '19

Hi , I have a query regarding Kamailio open source SIP server. Does it have any release, where its supports the feature of IMS supporting VOLTE with real commercial mobile phones(IMSI based) ? If so, what is the release number ? Please also let me know about the supported mobile phones list. BR, Shubhendu -- _Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._

2 2

Contact header requirement
by David Villasmil 02 Dec '19

02 Dec '19

Hello guys, I'm connecting to a provider who requires the contact contains the proxy's IP... I know this is weird and totally off-rfc... I setup the topoh module and used modparam("topoh", "mask_ip", "[PROXY_IP]:5060") but this makes the contact: Contact: <sip:[PROXY_IP]:5060;line=sr-Z213nshDqt98-2qiIoT8GFf. y8PWype.SRy0nRo3nCT*> I know this is how it works, but they don't like the additional ";line=...." Has anyone faced this kind of thing? is there any other way to set this to something like: Contact: <sip:[PROXY_IP]:5060> that won't mess everything? Thanks all, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 7

$null + 0 is not int
by Juha Heinanen 02 Dec '19

02 Dec '19

Why is it that $null + 0 is 0 but not int? What is it? -- Juha $avp(test) = $null + 0; if (is_int("$avp(test)")) INFO(test $avp(test) IS int); else INFO(test $avp(test) IS NOT int); Dec 1 02:46:01 salmon /usr/bin/sip-proxy[10138]: INFO: test 0 IS NOT int

2 5

Kamailio World 2020 - Call For Presentations
by Daniel-Constantin Mierla 02 Dec '19

02 Dec '19

Hello, I would like to announce that Call for Presentations at Kamailio World 2020 is now open. You can submit your proposal or see more details at: - https://www.kamailioworld.com/k08/call-for-speakers/ The 8th edition of the event takes place again in Berlin, Germany, during April 27-29, 2020. Expect over 150 participants, developers and community members as well as representatives of other popular open source VoIP projects such as Asterisk or FreeSwitch. Looking forward to meeting many of you there! 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

1 0

Possible conflict between t_newtran and setflag for ACC
by Patrick Wakano 01 Dec '19

01 Dec '19

Hello list, Hope you all doing well! I am using the ACC module and using the setflag() function as done in several examples. It works fine. However, I've added the t_newtran() function almost in the begging of the INVITE handler to help the retransmission detection and after that I noticed the ACC was not saving anything in DB. So after debugging I discovered that if I call the t_newtran() before setting the ACC flags, the module will not save the calls in DB, but if I call it after setting the ACC flags, it works.... So my question is, is this a bug or it is a expected side effect so when one is using t_newtran you must be careful and set all your transaction flags before? (ACC are the only transaction flags I am using so can't tell if other modules have the same problem) This is happening in Kamailio 5.2.2. Thank you! Kind regards, Patrick Wakano

3 6

tcp limit, too many open files
by David Villasmil 01 Dec '19

01 Dec '19

Hello guys, I'm facing an issue where, if i start kamailio as root, all works properly, but if i start it as kamailio, i get hundreds of: handle_new_connect(): error while accepting connection(24): Too many open files My limits is like: kamailio soft core unlimited kamailio soft data unlimited kamailio soft fsize unlimited kamailio soft memlock unlimited kamailio soft nofile unlimited kamailio soft rss unlimited kamailio hard stack 240 kamailio soft cpu unlimited kamailio soft nproc unlimited kamailio soft as unlimited kamailio soft priority -11 kamailio soft locks unlimited kamailio soft sigpending unlimited kamailio soft msgqueue unlimited kamailio soft nice -11 kamailio soft nofile 65536 kamailio hard nofile 65536 root soft core unlimited root soft data unlimited root soft fsize unlimited root soft memlock unlimited root soft nofile unlimited root soft rss unlimited root hard stack 240 root soft cpu unlimited root soft nproc unlimited root soft as unlimited root soft priority -11 root soft locks unlimited root soft sigpending unlimited root soft msgqueue unlimited root soft nice -11 root soft nofile 65536 root hard nofile 65536 I can't figure out why this is happening... Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

7 17

Kamailio as SBC for Microsoft Teams
by Henning Westerholt 29 Nov '19

29 Nov '19

Hello, I have created a "How-To" blog post on using Kamailio as "session border controller" for Microsoft Teams Direct Routing: https://skalatan.de/de/blog/kamailio-sbc-teams You can this way use all the existing possibilities that Kamailio provide to interact with MS Teams as well. Best regards, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

7 10

Adding "q" value for the main branch
by Maxim M 28 Nov '19

28 Nov '19

Maxim Malygin <maxim.malygin(a)gmail.com> 17:08 (5 минут назад) кому: sr-users Hello, I need to form list of Contact headers with "q" parameters for "302 Moved Temporary". I do something like this: ... # $xavp(contacts) - contains list of contact URIs (uri) and Q values (q) $var(i) = 0; $var(num) = $cnt($xavp(contacts)); while($var(i) < $var(num)) { if($var(i) == 0) { # How to add q value to this branch? $ru = $xavp(contacts[$var(i)]=>uri); } else { append_branch("$xavp(contacts[$var(i)]=>uri)", "$xavp(contacts[$var(i)]=>q)"); } } send_reply("302", "Moved Temporarily"); ... Is it possible to set "q" value for the main branch? For additional branches I do it via append_branch("uri", q_value) but I didn't find any way to do it for the main branch pointed via $ru/$du. The pseudo-variable "$branch" also doesn't allow me to access the main branch. Thanks, Maxim

1 0

Get totally rid of H.264 SDP
by Kjeld Flarup 28 Nov '19

28 Nov '19

Hi I have a door telephone, with GSM fallback. Since telephone operators are going away from SS7, more and more traffic goes SIP. Thus we now have some numbers where we can see that doing an invite with H.264 causes them to fail. I have tried to remove coded 96 with sdp_remove_codecs_by_id, but it still leaves most of the video SDP in the invite. SDP received by kamailio Media Description, name and address (m): video 6100 RTP/AVP 96 Media Attribute (a): rtcp:6101 IN IP4 192.168.237.239 Media Attribute (a): rtpmap:96 H264/90000 Media Attribute (a): fmtp:96 profile-level-id=42000d;packetization-mode=1 Media Attribute (a): sendrecv After sdp_remove_codecs_by_id I send this. But without "96" Media Description, name and address (m): video 6534 RTP/AVP Media Attribute (a): sendrecv Media Attribute (a): rtcp:6535 Media Attribute (a): ice-ufrag:35ZngQdL Media Attribute (a): ice-pwd:DG6kUbcYwBFKNS6YzCG27wRlG4 Media Attribute (a): candidate:r9KO0wNPMTGktklS 1 UDP 2130706431 192.168.237.15 6534 typ host Media Attribute (a): candidate:r9KO0wNPMTGktklS 2 UDP 2130706430 192.168.237.15 6535 typ host My problem is that the door telephone does not do an ACK when it gets the 200 OK My suspicion is that it does not understand the video response Media Description, name and address (m): video 0 RTP/AVP 0 Media Attribute (a): ice-ufrag:35ZngQdL Media Attribute (a): ice-pwd:DG6kUbcYwBFKNS6YzCG27wRlG4 Media Attribute (a): sendrecv Is there a way to remove the whole video part and not just the coded id? It is not an option to disable video in the door telephone, as we need the video. It is just in the fallback is must be removed. -- --------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

1 0

Kamailio does not detect message
by Adesh Pandey 28 Nov '19

28 Nov '19

Hi, I have below scenario: User registers on Kamailio server, calls to a number which is forwarded to asterisk through dispatcher, asterisk server answers and sends a text message to the caller using sendText but first message is delivered to client with message "501 Not Implemented" but in the subsequent calls asterisk shows that the text was delivered but kamailio did not detect the message. Guys, any help would be appreciated, I am using the below mentioned config. https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg Kamailio version: 5.2

1 0

Kamailio developers meeting - follow up remarks
by Daniel-Constantin Mierla 27 Nov '19

27 Nov '19

Hello, last week we had the 2nd annual Kamailio Developers Meeting hosted by Sipgate in Dusseldorf, Germany: * https://www.kamailio.org/w/developers-meeting/ 16 people were at the event in various roles. Giacomo Vacca published on his blog a good summary of what happened there: * https://www.giacomovacca.com/2019/11/my-notes-on-kamailio-developer-meeting… This year we had a lot of discussions, as well as work done on multiple planes, not only Kamailio code. So I am trying to list here some of the conclusions for future development, the technical aspects of the meeting, so everyone is aware and can provide feedback. 1) Effective work was done on: * kamailio code * kamailio rpm packaging * kamailio tools (kamctl) * kamailio release process * kamailio project keys (to be used to sign the packages) 2) Documentation 2.a) Wiki * it was somehow a rough consensus to move the wiki content to github, along with changing the format from dokuwiki markdown to the standard/github markdown. This should enable people to make pull requests so developers or community members can review and aprove new content. It also makes it easier to contribute using existing github account, now the kamailio.org/wiki is requiring to make a dedicated account, which many prefer not to do it. * the presentation can be done either by using mkdocs to generate html files hosted on kamailio.org or using the github provided wiki portal. 2.b) Docs for variables and transformations * there was a proposal to move them in the documentation the modules that export them, there are pros and cons, needs more discussions. Now they are in the wiki, so this probably has to resume after deciding on 2.a). 3) Kamailio Modules 3.a) replication (dmq) - several participants discussed about negotiations between nodes to take active role on some cases (e.g., active dialogs) 3.b) api integration - quite some interest in JSON-based API routing, concluding in extending rtjson to cover more use cases 3.c) security - have options to restrict the use of TLS1.3 or newer 4) Kamailio Releases * v5.3.2 was released during the event, allowing to document the process * work to automatize the process is planned, then eventually assing teams for takeing cares of releases from specific branches 5) Kamailio Testing * existing docker-based testing framework should be extended and integrated in CD/CI pipeline 6) Kamailio packages 6.a) rpms * rpm.kamailio.org has been prepared and is expected to take over the opensuse build service for building rpms and hosting them. Expected to provide support for hosting many kamailio versions in the same release series so one can do downgrade to older releases. Also, there is work in progress to provide nightly builds. 6.b) debs * work is planned to offer many kamailio versions in the same release series 7) Kamailio tools * kamctl/kamdbctl should be obsoleted in favor of kamcli, which offers a better framework for input validation and output formatting, as well as better portability, no longer depending on shell interpreter 8) Various discussions * kemi exports from C point of view and how to combine the documentation for modules and their kemi exports * how to make kamaiio friendlier in virtualized environments (ended up in the need of making the use of advertised address a bit more dynamic) * project organizatoric topics - to be approached separately * next events - Fosdem - someone should submit a proposal to present about Kamailio 9) Long term goals We speak here more or less about Kamailio 6.0 ... * change the behaviour of the native config interpreter to be consistent with the other programming languages in terms of handling the response code (change what is now: the evaluation of negative value to false and positive value to true and the hidden return 0 to exit) * make the pool of processes more generic, so they can handle traffic from more sockets (being sip traffic or something else) -- this should make better use of resources, as some sockets might be less busy that others I hope I covered the important topics, if I remember something else, I will reply on this thread. Or maybe other participants can contribute missing topics. Should anyone have comments or suggestions on the above topics, or new ones, let's discuss on sr-users because it impacts the long term use of Kamailio (sr-dev is cc-ed now for notification purposes). Overall, there were 2 very intensive days, however in a friendly and relaxed environment offered by Sipgate. Extremely useful discussions, not only about Kamailio, but about RTC ecosystem. At the evening social network event event, a couple of external people joined, some of them presenting very interesting use cases of Kamailio. Many thanks to all participants that allocated time and resources to come to Dusseldorf, as well as to the companies that covered expenses for participants. 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

2 2

Kamailio xavp_contact - Segmentation Fault
by Ilie Soltanici 27 Nov '19

27 Nov '19

Hello, Kamailio is crashing when i'm trying to set the parameter; modparam("tm|usrloc", "xavp_contact", "ulattrs") That's happening with Kamailio 5.3.1 and 5.2.5 too. Crash is happening when Im register 2 devices with the same extension. This is the core dump: [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib64/libthread_db.so.1". Core was generated by `/usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /usr/local/etc/ka'. Program terminated with signal 11, Segmentation fault. #0 0x000000000057cca2 in xavp_get_internal (name=0x8, list=0x0, idx=0, prv=0x0) at core/xavp.c:288 288 core/xavp.c: No such file or directory. (gdb) bt #0 0x000000000057cca2 in xavp_get_internal (name=0x8, list=0x0, idx=0, prv=0x0) at core/xavp.c:288 #1 0x0000000000581869 in xavp_insert (xavp=0x0, idx=0, list=0x0) at core/xavp.c:752 #2 0x00007f05be0ee3a1 in ki_t_next_contacts (msg=0x7f05cb76c218) at t_serial.c:552 #3 0x00007f05be0f09e8 in t_next_contacts (msg=0x7f05cb76c218, key=0x0, value=0x0) at t_serial.c:756 #4 0x0000000000434d41 in do_action (h=0x7ffe93c9cad0, a=0x7f05cb642250, msg=0x7f05cb76c218) at core/action.c:1067 #5 0x00000000004418e8 in run_actions (h=0x7ffe93c9cad0, a=0x7f05cb642250, msg=0x7f05cb76c218) at core/action.c:1572 #6 0x0000000000441fa9 in run_actions_safe (h=0x7ffe93c9de90, a=0x7f05cb642250, msg=0x7f05cb76c218) at core/action.c:1636 #7 0x000000000065734e in rval_get_int (h=0x7ffe93c9de90, msg=0x7f05cb76c218, i=0x7ffe93c9cf78, rv=0x7f05cb642570, cache=0x0) at core/rvalue.c:912 #8 0x000000000065b8fe in rval_expr_eval_int (h=0x7ffe93c9de90, msg=0x7f05cb76c218, res=0x7ffe93c9cf78, rve=0x7f05cb642568) at core/rvalue.c:1910 #9 0x000000000065bd51 in rval_expr_eval_int (h=0x7ffe93c9de90, msg=0x7f05cb76c218, res=0x7ffe93c9d42c, rve=0x7f05cb642c98) at core/rvalue.c:1918 #10 0x0000000000434807 in do_action (h=0x7ffe93c9de90, a=0x7f05cb645028, msg=0x7f05cb76c218) at core/action.c:1043 #11 0x00000000004418e8 in run_actions (h=0x7ffe93c9de90, a=0x7f05cb637e48, msg=0x7f05cb76c218) at core/action.c:1572 #12 0x0000000000431767 in do_action (h=0x7ffe93c9de90, a=0x7f05cb583858, msg=0x7f05cb76c218) at core/action.c:691 #13 0x00000000004418e8 in run_actions (h=0x7ffe93c9de90, a=0x7f05cb574750, msg=0x7f05cb76c218) at core/action.c:1572 #14 0x0000000000442071 in run_top_route (a=0x7f05cb574750, msg=0x7f05cb76c218, c=0x0) at core/action.c:1657 #15 0x00000000005874a9 in receive_msg (buf=0xa6ec00 <buf.6868> "OPTIONS sip:10@192.168.0.231:5060 SIP/2.0\r\nVia: SIP/2.0/UDP 192.168.0.231;branch=z9hG4bKd8a9.1b3abe656532065414a2f35a8916008d.1\r\nVia: SIP/2.0/UDP 192.168.0.131:5060;received=192.168.0.131;branch=z9hG"..., len=694, rcv_info=0x7ffe93c9e4d0) at core/receive.c:341 #16 0x000000000047bf81 in udp_rcv_loop () at core/udp_server.c:541 #17 0x0000000000424f9d in main_loop () at main.c:1669 #18 0x000000000042c688 in main (argc=13, argv=0x7ffe93c9ea18) at main.c:2710 Thank you.

2 1

Kamailio Parrallel Forking
by Ilie Soltanici 27 Nov '19

27 Nov '19

Hello, I'm trying to use parallel forking if one extension is registered from multiple devices. (using WebSockets) This is the related code from kamailio.cfg: ------------------------------------------------------------------------------------------------------------------------------------ if (!t_load_contacts()) { xlogl("L_WARN", "Error loading contacts for $rU\n"); sl_send_reply("500", "Server Internal Error (Code:$cfg(line))"); exit; } else { xlogl("L_INFO", "Contacts loaded for $rU\n"); } if (!t_next_contacts()) { xlogl("L_INFO", "t_next_contacts - Only one contact found for $rU, calling\n"); } else { xlogl("L_INFO", "t_next_contacts - Multiple contacts found, parallel forking\n"); } route(RELAY); ------------------------------------------------------------------------------------------------------------------------------------ I noticed that on Kamailio 5.2.5 - this code is working fine, and Kamailio is sending an INVITE to both Devices.But, on Kamailio 5.3.1 - that's not working anymore. For some reason, kamailio is sending just one INVITE to one device with wrong URI. The URI is set to the second device instead of the correct one. Was something changed in the t_next_contacts function related to this? Thank you

1 0

Kamailio 5.2.3 segfaulting
by Daniel Greenwald 27 Nov '19

27 Nov '19

I'm seeing this in the log and kamailio is crashing every hour or so: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 33/var/log/kamailio/kamailio.log Any ideas? Should I emergency upgrade to 5.3.1?

2 12

No Audio using RTP Proxy
by Sujit Roy 27 Nov '19

27 Nov '19

Hello I am facing a problem as below. Please suggest for the work around. My call flow is like this. SIP Gateway-1 (IP x.179) -> SIP Gateway-2 ( IP x.177) -> Kamalio+RTPProxy So when the call arrives at Kamalio+RTPProxy, i m getting below in log. Nov 26 23:25:31 rtpproxy[18508]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 1b7c870763616c6c15fff410@ 192.168.100.177, tag 1aa18fc201a68168;1 requested, type strong Nov 26 23:25:31 rtpproxy[18508]: INFO:1b7c870763616c6c15fff410@ 192.168.100.177:rtpp_command_ul_handle: new session on IPv4 port 15920 created, tag 1aa18fc201a68168;1 Nov 26 23:25:31 rtpproxy[18508]: INFO:1b7c870763616c6c15fff410@192.168.100.177:rtpp_stream_prefill_addr: pre-filling caller's RTP address with 192.168.100.177:27360 Nov 26 23:25:31 rtpproxy[18508]: INFO:1b7c870763616c6c15fff410@ 192.168.100.177:rtpp_stream_prefill_addr: pre-filling caller's RTCP address with 192.168.100.177:27361 But x.177 is working on signalling mode only ( Not routing Media ) . As a result, i m not getting any voice from IP x.179 What can be done to change the caller's RTP address to x.179 in RTPProxy ? Thanks -- Regards =================== Sujit Roy

2 1

Encode host ip in Contact header for semi-stateless setup
by Daniel Greenwald 27 Nov '19

27 Nov '19

At kamailio world 2017, Jan Lorenz of Pascom explained how they use kamailio in a semi-stateless manner and encode the asterisk's IP into the contact header on the way out so that it can be decoded and used to route properly on the way back without Record-Routing. I would like to create such a setup. I'm looking for pointers if there is a particular module/method that would be ideal to to use for this. Any other gotcha's to consider in such a setup. My plan is to keep the transaction on the same kamailio by not altering the VIA but allow the subsequent in-dialog transactions to potentially hit another kamailio by changing the contact to a shared SRV record (and encoding the freeswitch box's IP into it for later retrieval.) Thanks in advance. Daniel Greenwald

4 4

dialog module: Does dlg_manage() have to be called for all requests?
by George Diamantopoulos 27 Nov '19

27 Nov '19

Hello all, I've been reading through the dialog module docs again, and there seems to be a discrepancy in what's suggested in the docs. In the intro, it is stated that 'To create the dialog associated with an initial INVITE request, execute the function “dlg_manage()"'. Later on, in section "7.10" the following example code is provided: request_route { ... if(is_method("INVITE") && !has_totag()) { $dlg_ctx(timeout_route) = "DLGTIMEOUT"; $dlg_ctx(timeout_bye) = 1; } dlg_manage(); ... } In this example code, dlg_manage seems to be called for all requests, as it is not dependent on the conditional that only requests with no to-tag are to be handled. Previously, Daniel had suggested to me in person that one also run dlg_manage() for CANCELs, and in-dialog reINVITEs, BYEs and ACKs to mitigate some buggy corner cases. Is this still the case? What do you guys and gals do in production? Have you had to revise how you use dlg_manage in terms of calling it for initial INVITEs only vs all requests? Thanks! BR, George

5 6

Modify SIP Message Header and Message Body parameter
by Sujit Roy 26 Nov '19

26 Nov '19

Hello Please see below SIP Message SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.1681.101:5060;branch=z9hG4bK5a61798d7c850d39;rport=5060 From: <sip:summona@ 192.1681.101>;tag=51a0dc1e620de16d To: <sip:101880171234567@ 192.1681.102> Call-ID: 6fd9b9d563616c6c15f2d7fa@ 192.1681.101 CSeq: 21701 INVITE *Server: kamailio (5.1.9 (x86_64/linux))* Content-Length: 0 *1. I want to Change the Server Parameter . How to do it ? * SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.1681.101:5060;rport=5060;branch=z9hG4bK5a61798d7c850d39 Record-Route: <sip: 192.1681.102;lr> From: <sip:summona@ 192.1681.101>;tag=51a0dc1e620de16d To: <sip:101880171234567 @ 192.1681.102>;tag=as728725b3 Call-ID: 6fd9b9d563616c6c15f2d7fa@ 192.1681.101 CSeq: 21701 INVITE Server: VOS3000 V2.1.4.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 600;refresher=uas *Contact: <sip:1018801751481417@104.216.39.66:5060 <http://sip:1018801751481417@104.216.39.66:5060>>* Content-Type: application/sdp Require: timer Content-Length: 289 v=0 o=root 912643161 912643161 IN IP4 192.1681.104 s=VOS3000 V2.1.4.0 c=IN IP4 192.1681.104 t=0 0 m=audio 10784 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:230 a=sendrecv *a=nortpproxy:yes* 2. I Want to hide the Contact Parameter and a= nortpproxy:yes ==> How to do it as well ? Thanks in advance. -- Regards =================== Sujit Roy

2 1

Beginner Help: Relay vs Forward
by Michael Iedema 26 Nov '19

26 Nov '19

Hi all, I am trying to place Kamailio between several SIP services which are differentiated by port and by the type of message they handle. To be concrete, here are the services: Originator: - reachable on port 5060 - originates all traffic to the services Service A: - reachable on port 5061 - handles REGISTER Service B: - reachable on port 5062 - handles INVITE Service C: - reachable on port 5063 - handles MESSAGE The services, originator and Kamailio instance may all be run on the same IP or may be split onto different addresses. Previously, the originator needed to know the address and port of each service. Now I want to make Kamailio responsible for that task so it can be done more intelligently. To start though I simply want to get traffic flowing between the services and this is where I’m struggling to climb the learning curve. In my test setup the originator and all services are on the same IP with only Kamailio running on a different IP. Address 1 Address 2 ======================== Originator <--> Kamailio:6060 | | Service A <-------> | Service B <-------> | Service C <-------> | My first attempt at the logic looks like this: # HOST DEFINITIONS #!substdef "/HOST_ORIGINATOR/192.168.86.110/" #!substdef "/HOST_REGISTER/192.168.86.110/“ #!substdef "/HOST_INVITE/192.168.86.110/" #!substdef "/HOST_MESSAGE/192.168.86.110/" # PORT DEFINITIONS #!substdef “/PORT_ORIGINATOR/5060/" #!substdef "/PORT_REGISTER/5061/“ #!substdef "/PORT_INVITE/5062/" #!substdef "/PORT_MESSAGE/5063/" # ROUTE LOGIC route { # ORIGINATOR sent us something if ( $(ct{nameaddr.uri}{uri.port}) == HOST_ORIGINATOR ) { xlog("L_INFO", "ORIGINATOR: sent $rm\n”); if (is_method("REGISTER")) { $du = "sip: HOST_REGISTER:PORT_REGISTER"; xlog("L_INFO", "ORIGINATOR -> REGISTER: Destination URI is now $du\n"); t_relay(); exit; } } … #### END SNIPPET #### My very newbie questions are: - Am I even on the right track here? I understand that t_relay() establishes a stateful dialog handler between the originator and register services. - Can I accomplish this bi-directional path statelessly with forward()? - Do I need to add logic to handle the messages from the REGISTER service back to ORIGINATOR? This config currently fails because of "no corresponding socket found” when the relay triggers. I believe this is because I’m running the Kamailio instance in a Docker container so my first step today is to set up a native install. I also looked at the dispatcher module which seems very close to what I want to do. Maybe that’s a more natural way to go? Anyway, apologies for the lengthy explanation. I hope it is at least clear what I intend to do. I will gladly research and do trial and error to solve this but need to know I’m going down roughly the right path. I’m not sure what I should even be googling even of the time :-) Thanks as always for your time and the great software! Regards, -Michael

4 5

Re-use TCP connections on different transactions
by Joel Serrano 26 Nov '19

26 Nov '19

Hello, I'm setting up a Kamailio instance behind a TCP load balancer (with proxy protocol and NAT routing: meaning Kam stays in the flow all the time). I've managed to get working almost everything we need for our service, except for one thing, and that is for Kam to use existing connections for subsequent transactions: Following this example: [image: image.png] EXT & INT represent the external and internal interface of a LB between the UAC and Kamailio, using TLS on both legs and proxy protocol. Transaction 1: INVITE, 100, 180, 183, 200 OK UAC 1.1.1.1:1111 -> 2.2.2.2:443 (EXT) 3.3.3.3:3333 (INT) -> 7.7.7.7:5060 (Kamailio) Transaction 2: ACK UAC 1.1.1.1:1112 -> 2.2.2.2:443 (EXT) 4.4.4.4:4444 (INT) -> 7.7.7.7:5060 (Kamailio) Transaction 3: BYE Kam 7.7.7.7:5060 -> 3.3.3.3:3333 (INT) 2.2.2.2:443 (EXT) -> 1.1.1.1:1111 UAC My problem is with Transaction 3. In this case the BYE is originated by the callee, and Kam has to send it to the caller. As the TCP load balancer is between Kam and the UAC, Kam has to send it to the LB so then the LB can forward it back to the UAC. This works well for msgs that belong to the same transaction (INVITE, 100, 180, 183, 200 OK) but it fails when they don't belong to the same transaction. Thanks to the newly added $tcp(c_si) and $tcp(c_sp) pseudovars, I can save the internal IP:Port of the LB, so I can send stuff later to it, my problem is that Kam doesn't seem to allow this? On the original INVITE, I use the following to save where I have to reach the UAC: add_contact_alias("$tcp(c_si)", "$tcp(c_sp)", "tls"); Then, handle_ruri_alias() will take care of setting $du to the correct (internal LB) IP:Port so I can reach the UAC, this works. My problem is that Kamailio doesn't identify that there is a valid existing TLS connection still up (from the INVITE), and tries to create a new one (and this obviously doesn't gives all sorts of problems). So when I run handle_ruri_alias(), and $du is set to 3.3.3.3:3333 (from the example above), instead of using the existing connection, Kamailio tries to create a new one. I have a log statement right before with the result of tcp_conid_state(1) (the connid is 1 for this connection) and the $rc is 1 (Connection is OK), but when I tell Kamailio it has to use it I get this in the logs: DEBUG: {1 11726467 BYE gqR1qqNK8B} <core> [core\/tcp_main.c:2060]: tcp_send(): no open tcp connection found, opening new one"} And then the problems begin... I have tried playing around with: tcp_reuse_port tcp_connection_match But no luck..! I also thought it could be a problem of the connection being created on one worker, and a different worker handling BYE transaction, so tested with children=1 and tcp_children=1, but still same problem. A more detailed log: In blue my log statement checking for the status of conid "1", in red Kam not being able to find it, although it exists (as validated by tcp_conid_state(), and even in netstat I can see the connection established). In this log, 35.191.0.66:60271 would be the equivalent of 3.3.3.3:3333 and 104.175.176.242:28157 would be 1.1.1.1:1111 from the example above. ... "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/tcp_main.c:1657]: _tcpconn_find(): found connection by id: 1"} "message":" NOTICE: {1 11727734 BYE 5-LX4GdI9X} <script>: JOEL TEST New request - M=BYE TCP STATUS:1 ID=5-LX4GdI9X"} ... "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} nathelper [nathelper.c:1144]: handle_ruri_alias(): setting dst_uri to <sip:35.191.0.66:60271;transport=tls>"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} nathelper [nathelper.c:1166]: handle_ruri_alias(): rewriting r-uri to <sip:linphone@104.175.176.242:28157;transport=tls>"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tm [t_lookup.c:1328]: t_newtran(): msg (0x7f3c884259d0) id=534\/18664 global id=532\/18664 T start=0xffffffffffffffff"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tm [t_lookup.c:497]: t_lookup_request(): start searching: hash=63496, isACK=0"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tm [t_lookup.c:455]: matching_3261(): RFC3261 transaction matching failed - via branch [z9hG4bK808f.eee2444f92a02cb33e1b7a21f20bc6bb.0]"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tm [t_lookup.c:675]: t_lookup_request(): no transaction found"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tm [t_hooks.c:336]: run_reqin_callbacks_internal(): trans=0x7f3c83b8c598, callback type 1, id 0 entered"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/crypto\/md5utils.c:67]: MD5StringArray(): MD5 calculated: 3071029feb05962b26b53a9664a14210"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} siputils [checks.c:123]: has_totag(): totag found"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} rr [loose.c:1095]: check_route_param(): route params checking against [;r2=on;lr=on;ftag=Eb~TbdfTA;did=cab.01e2;nat=yes] (orig: [r2=on;lr=on;ftag=Eb~TbdfTA;did=cab.01e2;nat=yes])"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} rr [loose.c:1101]: check_route_param(): params are <;r2=on;lr=on;ftag=Eb~TbdfTA;did=cab.01e2;nat=yes>"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} siputils [checks.c:123]: has_totag(): totag found"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/msg_translator.c:161]: check_via_address(): (198.1.54.228, 198.1.54.228, 0)"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/tcp_main.c:2060]: tcp_send(): no open tcp connection found, opening new one"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/ip_addr.c:229]: print_ip(): tcpconn_new: new tcp connection: 35.191.0.66"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/tcp_main.c:1242]: tcpconn_new(): on port 60271, type 3"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/tcp_main.c:1561]: tcpconn_add(): hashes: 1446:2350:0, 5"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tls [tls_server.c:199]: tls_complete_init(): completing tls connection initialization"} ... So time to seek help from the community, any suggestions/ideas/comments? Sorry if all this sounds confusing, I've tried my best to put in text the whole scenario in and "understandable" way... Is this even doable? Thanks, Joel.

3 10

How to enforce REGISTER with AKAv1 (IMS)
by Moshe A 26 Nov '19

26 Nov '19

Hello, How can we enforce REGISTER with AKAv1? Here our setup 1. kamailio version 4.3.4 for IMS 2. Boghe IMS client v1.0.58.550 3. scscf.cfg is configure with: #!define REG_AUTH_DEFAULT_ALG "AKAv1-MD5" 4. Set Authentication Schemes in Private User Identity - IMPI for user bob to: Digest-AKAv1 (3GPP) results: First REGISTER fails with 401 Unauthorized Second REGISTER 200 OK, but the REGISTER is MD5 and not AKAv1 Attached is tcpdump. By the way: in which DB should we find Authentication Schemes? can't find it in hss_db, kamailio and not in scscf. Thank you, Moshe.

1 0

call not going to end point
by Gaurav Bmotra 26 Nov '19

26 Nov '19

hi i m using kamailio 5.4 dev Docker image when i try to call form calientA to clientB i got (Temporarily Unavailable (code: 480)) i m using freeswitch as media server so the flow and structure should be like this ClientA---->Kamailio------>FS---->Kamailio>---->ClientB but issue is (what i m getting ) ClientA---->Kamailio------>FS---->Kamailio>-------->FS-------->kamailio--------->FS IT CREATE THE CALL LOOP b/w KAMAILIO AND FREESWITCH *MY CONFIG FILE IS ATACHED * in logs i m getting -------------------------------------------------------- Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:create_twinlistener: can't bind to the IPv4 port 38902: Address not available (99) Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:rtpp_command_ul_handle: can't create listener Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:rtpc_doreply: sending reply "E72\n" Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:get_command: received command "18_4 Uc0,8,101 9857c00a-8947-1238-c6a7-12590b724429 3.82.96.70 22082 jKrKymKNg480p;1" Nov 24 10:25:36 ip-172-20-61-37 local0.info rtpproxy[6]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 9857c00a-8947-1238-c6a7-12590b724429, tag jKrKymKNg480p;1 requested, type strong Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:create_twinlistener: can't bind to the IPv4 port 35888: Address not available (99) Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:rtpp_command_ul_handle: can't create listener Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:rtpc_doreply: sending reply "E72\n" Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:get_command: received command "20_4 Uc0,8,101 98589959-8947-1238-c6a7-12590b724429 3.82.96.70 29846 m5a51aNvapN6D;1" Nov 24 10:25:36 ip-172-20-61-37 local0.info rtpproxy[6]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 98589959-8947-1238-c6a7-12590b724429, tag m5a51aNvapN6D;1 requested, type strong Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:create_twinlistener: can't bind to the IPv4 port 37706: Address not available (99) Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:rtpp_command_ul_handle: can't create listener Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:rtpc_doreply: sending reply "E72\n" Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:get_command: received command "13_4 Uc0,8,101 985962c1-8947-1238-c6a7-12590b724429 3.82.96.70 29432 pQXp50p3471BN;1" Nov 24 10:25:36 ip-172-20-61-37 local0.info rtpproxy[6]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 985962c1-8947-1238-c6a7-12590b724429, tag pQXp50p3471BN;1 requested, type strong Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:create_twinlistener: can't bind to the IPv4 port 45078: Address not available (99) Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:rtpp_command_ul_handle: can't create listener Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:rtpc_doreply: sending reply "E72\n" Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:get_command: received command "16_5 Uc0,8,101 985a3ce3-8947-1238-c6a7-12590b724429 3.82.96.70 25096 r9F88praZSeHc;1" Nov 24 10:25:36 ip-172-20-61-37 local0.info rtpproxy[6]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 985a3ce3-8947-1238-c6a7-12590b724429, tag r9F88praZSeHc;1 requested, type strong Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:create_twinlistener: can't bind to the IPv4 port 58376: Address not available (99) Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:rtpp_command_ul_handle: can't create listener Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:rtpc_doreply: sending reply "E72\n" -------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- *here is my config file * #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ACCDB #!define WITH_NAT # # Kamailio (OpenSER) SIP Server v5.4 - default configuration script # - web: https://www.kamailio.org # - git: https://github.com/kamailio/kamailio # # Direct your questions about this file to: <sr-users(a)lists.kamailio.org> # # Refer to the Core CookBook at https://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Note: the comments can be: # - lines starting with #, but not the pre-processor directives, # which start with #!, like #!define, #!ifdef, #!endif, #!else, #!trydef, # #!subst, #!substdef, ... # - lines starting with // # - blocks enclosed in between /* */ # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To use RTPEngine (instead of RTPProxy) for nat traversal execute: # - define WITH_RTPENGINE # - install RTPEngine: https://github.com/sipwise/rtpengine # - start RTPEngine: # rtpengine --listen-ng=127.0.0.1:2223 ... # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To block 401 and 407 authentication replies execute: # - define WITH_BLOCK401407 # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes for each UDP socket * - value inherited by tcp_children and sctp_children when not set explicitely */ children=8 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes /* number of SIP routing processes for all TCP/TLS sockets */ # tcp_children=8 /* uncomment the next line to disable the auto discovery of local aliases * based on reverse DNS on IPs (default on) */ # auto_aliases=no /* add local domain aliases */ # alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to * bind on a specific interface/port/proto (default bind on all available) */ # listen=udp:10.0.0.10:5060 listen=udp:LOCAL_IP:5091 advertise PUBLIC_IP:5091 /* life time of TCP connection when there is no traffic * - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 /* upper limit for TCP connections (it includes the TLS connections) */ tcp_max_connections=2048 #!ifdef WITH_TLS enable_tls=yes /* upper limit for TLS connections */ tls_max_connections=2048 #!endif ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ #!ifdef WITH_PSTN /* PSTN GW Routing * * - pstn.gw_ip: valid IP or hostname as string value, example: * pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" * * - by default is empty to avoid misrouting */ pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL /* VoiceMail Routing on offline, busy or no answer * * - by default Voicemail server IP is empty to avoid misrouting */ voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" loadmodule "dispatcher.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" #!ifdef WITH_RTPENGINE loadmodule "rtpengine.so" #!else loadmodule "rtpproxy.so" #!endif #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- #-------------------dispatcher params------------- modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/run/kamailio/kamailio_ctl") # ----- sanity params ----- modparam("sanity", "autodrop", 0) # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ # modparam("registrar", "max_contacts", 10) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT #!ifdef WITH_RTPENGINE # ----- rtpengine params ----- modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223") #!else # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") #!endif # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- /* ip ban htable with autoexpire after 5 minutes */ modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains #route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN #route(PSTN); route(DISPATCH); # user location service } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); $rcv_addr="sip:"+$si+":"+$sp; if(starts_with("$rcv_addr","$du")) { route(LOCATION); } else { route(RELAY); } exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { xnotice("RTF_DISPATCH: $rm $rU [$ci]"); if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("(^5)") or (t_branch_timeout() and !t_branch_replied())) { xnotice("RTF_DISPATCH: WARNING $du is broken and marked as inactive"); # ds_mark_dst("ip"); if(ds_next_dst()) { #xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } # Per SIP request initial checks route[REQINIT] { # no connect for sending replies set_reply_no_connect(); #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if($ua =~ "friendly-scanner|sipcli|sipvicious|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP request from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) { sl_reply_error(); } exit; } # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; #!ifdef WITH_RTPENGINE if(nat_uac_test("8")) { rtpengine_manage("replace-origin replace-session-connection"); } else { rtpengine_manage("trust-address replace-origin replace-session-connection"); } #!else if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } #!endif if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } if(isbflagset(FLB_NATB)) { # no connect message in a dialog involving NAT traversal if (is_request()) { if(has_totag()) { set_forward_no_connect(); } } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + if (starts_with("$rU", "00")) { strip(2); prefix("+"); } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies reply_route { if(!sanity_check("17604", "6")) { xlog("Malformed SIP response from $si:$sp\n"); drop; } } # Manage incoming replies in transaction context onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif } -- *Regards:* Gaurav Kumar

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How to get the audio codec used for each established call.
by Abdoul Osséni 25 Nov '19

25 Nov '19

Hello all, I try to save in the CDR the audio codec used for each established call. Can you help me? Regards Abdoul.I

3 4

SDP Modification
by Sujit Roy 25 Nov '19

25 Nov '19

Hello How can i change c= and o= in SDP using RTPProxy ? Please advise Thanks -- Regards =================== Sujit Roy

2 1

Passing media thru RTP Proxy
by Sujit Roy 25 Nov '19

25 Nov '19

Hello I have installed and configured both Kamailio and RTP Proxy and calls are going. But i m unable to force media to go thru RTP Proxy. How can i enforce media to go thru RTP Proxy ? Thanks -- Regards =================== Sujit Roy

3 2

Kamailio stopped relaying SIP text message from asterisk
by Adesh Pandey 25 Nov '19

25 Nov '19

Hi, I have been working on kamailio with following scenario: 1. Webrtc SIP Users register to Kamailio 2. Users initiate outgoing calls to some other numbers 3. Kamailio dispatches outgoing calls to Asterisk 4. Asterisk answers the call and then send a text message to the caller using SendText, which was working fine a few days back but all of sudden it has stopped working 5. At Asterisk end it is showing that message was delivered successfully but when I check SIP user logs, there is no message event. Please review my config file at following URL: https://gist.github.com/adeshpandey/2c338db7f5d992267f7415de0325e0b4 Developers, I also want to know if there is any way to log *text message* in kamailio server.

3 3

Unable to connect to sip server via proxy
by Ashwini Kumar 24 Nov '19

24 Nov '19

Hi I have two network interfaces on my system in which one interface is connected to my office WAN/LAN and other is connected to SIP private network connected to PSTN. Now, I want to access sip server using the internal IP of our office LAN as proxy. *192.168.6.2 (Office LAN IP as proxy) -> 10.50.185.2 (SIP)* So on machine 192.168.6.2, I setup kamailio as SIP proxy server so that our office LAN machine can connect to SIP server directly via SIP proxy. I used config defined in - *https://github.com/kamailio/kamailio/blob/master/misc/examples/mixed/kamailio-minimal-proxy.cfg* <https://github.com/kamailio/kamailio/blob/master/misc/examples/mixed/kamail…> and I changed the local domain alias with SIP IP -* 10.50.185.2*. When I try to connect to SIP via proxy using *linphone* (UDP) I got below errors in* /var/log/syslog* and shows *connection in progress* every time. Nov 23 15:15:49 ubantudesk /usr/sbin/kamailio[25784]: WARNING: {1 20 REGISTER ek82LguQ6z} sanity [sanity.c:585]: check_cl(): content length header missing in request Nov 23 15:15:53 ubantudesk /usr/sbin/kamailio[25785]: WARNING: {1 20 REGISTER ek82LguQ6z} sanity [sanity.c:585]: check_cl(): content length header missing in request Nov 23 15:15:57 ubantudesk /usr/sbin/kamailio[25784]: WARNING: {1 20 REGISTER ek82LguQ6z} sanity [sanity.c:585]: check_cl(): content length header missing in request Nov 23 15:16:01 ubantudesk /usr/sbin/kamailio[25785]: WARNING: {1 20 REGISTER ek82LguQ6z} sanity [sanity.c:585]: check_cl(): content length header missing in request Nov 23 15:16:05 ubantudesk /usr/sbin/kamailio[25784]: WARNING: {1 20 REGISTER ek82LguQ6z} sanity [sanity.c:585]: check_cl(): content length header missing in request Can you help me what I should do? -- *Thanks & Regards,* *Ashwini Kumar | Technology |* www.industrybuying.com | Ghitorni, Delhi - 110030 | India | È*+91-8802137782* | * ashwinikumar(a)industrybuying.com |

2 1

Avoid full table scan for dispatcher table on db_redis using ds_reload() from config script
by Joel Serrano 23 Nov '19

23 Nov '19

Hello, I'm trying out redis as db backend, and right now I only have dispatcher records there... Every so often, I do a ds_reload() from within Kam config script, and I see in logs: Nov 22 20:36:35 test COPS[25531]: WARNING: db_redis [redis_dbase.c:1098]: db_redis_perform_query(): performing full table scan on table 'dispatcher' while performing query After enabling debug logs: Nov 22 21:15:35 test COPS[26661]: DEBUG: db_redis [redis_dbase.c:1761]: db_redis_query(): querying prefix (table) 'dispatcher' Nov 22 21:15:35 test COPS[26661]: DEBUG: db_redis [redis_dbase.c:1811]: db_redis_query(): no columns given to build query keys, falling back to full table scan Nov 22 21:15:35 test COPS[26661]: DEBUG: <core> [db_res.c:119]: db_new_result(): allocate 56 bytes for result set at 0x7f0be4273898 Nov 22 21:15:35 test COPS[26661]: DEBUG: <core> [db_res.c:156]: db_allocate_columns(): allocate 40 bytes for result names at 0x7f0be4273938 Nov 22 21:15:35 test COPS[26661]: DEBUG: <core> [db_res.c:167]: db_allocate_columns(): allocate 20 bytes for result types at 0x7f0be42739c8 Nov 22 21:15:35 test COPS[26661]: WARNING: db_redis [redis_dbase.c:1098]: db_redis_perform_query(): performing full table scan on table 'dispatcher' while performing query Given the msg... "no columns given to build query keys, falling back to full table scan", I assume I'm missing a keys modparam.. but I don't know how to build it, and the docs (to me) seem confusing? https://kamailio.org/docs/modules/devel/modules/db_redis.html#db_redis.p.ke… Can anyone help me understand how to build the keys modparam for dispatcher table? Or if the reason for this is different, what do I need to do to avoid the full table scan and the warning in the log? Thanks, Joel.

1 1

Question about ims_ipsec_pcscf
by Woscek, Martin W. 22 Nov '19

22 Nov '19

Hi all, We are looking for information as to what the ims_ipsec_pcscf module was written for and tested against. What was the target UE/EPC/PCRF that Tsvetomir Dimitrov wrote the module towards? Has an emulated EPC been tested with this module? If so which one(s)? Thanks in advance, Martin

1 0

Cannot read an indexed xavp from Kemi-Lua
by Michel Pelletier 22 Nov '19

22 Nov '19

Hi, I am having trouble reading an indexed xavp from Kemi-Lua. The xavp contains 3 entries but I can only access the first one. The xavp is the one filled by the dispatcher module after a call to KSR.dispatcher.ds_select(group, 4) with the flags parameter set to 2. I can get the number of entries using KSR.pvx.xavp_child_get("ds_ctx", "cnt") which returns 3. I can also get the contents of the first element using, for example, KSR.pvx.xavp_child_get("ds_dst", "attrs"). But all my attempts at adding an index value using KSR.pvx.xavp_get() or KSR.pvx.xavp_child_get() return the value 'nil'; Below is the output of KSR.pvx.pv_xavp_print(): +++++ start XAVP list: (nil) ((nil)) (level=0) ----- end XAVP list: (nil) (level=0) +++++ start XAVP list: 0x7f94ce01f1f8 ((nil)) (level=0) *** (l:0 - 0x7f94ce01f1f8) XAVP name: ds_ctx XAVP id: 1762531321 XAVP value type: 6 XAVP value: <xavp:0x7f94ce01f150> +++++ start XAVP list: 0x7f94ce01f150 (0x7f94ce01f218) (level=1) *** (l:1 - 0x7f94ce01f150) XAVP name: cnt XAVP id: 7279890 XAVP value type: 1 XAVP value (int): 3 *** (l:1 - 0x7f94ce01f150) end ----- end XAVP list: 0x7f94ce01f150 (level=1) *** (l:0 - 0x7f94ce01f1f8) end *** (l:0 - 0x7f94ce01eaf8) XAVP name: ds_dst XAVP id: 1762531862 XAVP value type: 6 XAVP value: <xavp:0x7f94ce01ea30> +++++ start XAVP list: 0x7f94ce01ea30 (0x7f94ce01eb18) (level=1) *** (l:1 - 0x7f94ce01ea30) XAVP name: attrs XAVP id: 1835799177 XAVP value type: 2 XAVP value (str): duid=newjersey3;maxload=200 *** (l:1 - 0x7f94ce01ea30) end *** (l:1 - 0x7f94ce01e988) XAVP name: grp XAVP id: 7056653 XAVP value type: 1 XAVP value (int): 1 *** (l:1 - 0x7f94ce01e988) end *** (l:1 - 0x7f94ce01e8c8) XAVP name: uri XAVP id: 8122237 XAVP value type: 2 XAVP value (str): sip:123.123.123.123:5060 *** (l:1 - 0x7f94ce01e8c8) end ----- end XAVP list: 0x7f94ce01ea30 (level=1) *** (l:0 - 0x7f94ce01eaf8) end *** (l:0 - 0x7f94ce01edd0) XAVP name: ds_dst XAVP id: 1762531862 XAVP value type: 6 XAVP value: <xavp:0x7f94ce01ed08> +++++ start XAVP list: 0x7f94ce01ed08 (0x7f94ce01edf0) (level=1) *** (l:1 - 0x7f94ce01ed08) XAVP name: attrs XAVP id: 1835799177 XAVP value type: 2 XAVP value (str): duid=newjersey2;maxload=200 *** (l:1 - 0x7f94ce01ed08) end *** (l:1 - 0x7f94ce01ec60) XAVP name: grp XAVP id: 7056653 XAVP value type: 1 XAVP value (int): 1 *** (l:1 - 0x7f94ce01ec60) end *** (l:1 - 0x7f94ce01eba0) XAVP name: uri XAVP id: 8122237 XAVP value type: 2 XAVP value (str): sip:123.123.123.123:5060 *** (l:1 - 0x7f94ce01eba0) end ----- end XAVP list: 0x7f94ce01ed08 (level=1) *** (l:0 - 0x7f94ce01edd0) end *** (l:0 - 0x7f94ce01f0a8) XAVP name: ds_dst XAVP id: 1762531862 XAVP value type: 6 XAVP value: <xavp:0x7f94ce01efe0> +++++ start XAVP list: 0x7f94ce01efe0 (0x7f94ce01f0c8) (level=1) *** (l:1 - 0x7f94ce01efe0) XAVP name: attrs XAVP id: 1835799177 XAVP value type: 2 XAVP value (str): duid=newjersey1;maxload=200 *** (l:1 - 0x7f94ce01efe0) end *** (l:1 - 0x7f94ce01ef38) XAVP name: grp XAVP id: 7056653 XAVP value type: 1 XAVP value (int): 1 *** (l:1 - 0x7f94ce01ef38) end *** (l:1 - 0x7f94ce01ee78) XAVP name: uri XAVP id: 8122237 XAVP value type: 2 XAVP value (str): sip:123.123.123.123:5060 *** (l:1 - 0x7f94ce01ee78) end ----- end XAVP list: 0x7f94ce01efe0 (level=1) *** (l:0 - 0x7f94ce01f0a8) end ----- end XAVP list: 0x7f94ce01f1f8 (level=0) Regards, Michel Pelletier

1 0

kamailio call loop issue B/W kamailio and freeswitch
by Gaurav Bmotra 22 Nov '19

22 Nov '19

hi im using kamailio 5.3 as register server and using dispatcher i m dendign the calls to freeswitch .. ibut the issue is then freeswitch send back call to kamailio after processing media ... it should go back to client but it again send call to freeswitch ,, which create the loop and hangup the call with (Temporarily Unavailable (code: 480)) *what i want *client1------>Kamailio------>FS-----Kamailio----->Client2 *but I am getting this * client1------>Kamailio------>FS-----Kamailio----->FS----->kamailio----->FS -- here is my config file data --- #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ACCDB #!define WITH_NAT # # Kamailio (OpenSER) SIP Server v5.2 - default configuration script # - web: https://www.kamailio.org # - git: https://github.com/kamailio/kamailio # # Direct your questions about this file to: <sr-users(a)lists.kamailio.org> # # Refer to the Core CookBook at https://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Note: the comments can be: # - lines starting with #, but not the pre-processor directives, # which start with #!, like #!define, #!ifdef, #!endif, #!else, #!trydef, # #!subst, #!substdef, ... # - lines starting with // # - blocks enclosed in between /* */ # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To block 401 and 407 authentication replies execute: # - define WITH_BLOCK401407 # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes for each UDP socket * - value inherited by tcp_children and sctp_children when not set explicitely */ children=8 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes /* number of SIP routing processes for all TCP/TLS sockets */ # tcp_children=8 /* uncomment the next line to disable the auto discovery of local aliases * based on reverse DNS on IPs (default on) */ # auto_aliases=no /* add local domain aliases */ # alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to * bind on a specific interface/port/proto (default bind on all available) */ # listen=udp:10.0.0.10:5060 listen=udp:172.20.LOCAL_IP:5091 advertise 3.83.PUBLIC_IP:5091 /* life time of TCP connection when there is no traffic * - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 /* upper limit for TCP connections (it includes the TLS connections) */ tcp_max_connections=2048 #!ifdef WITH_TLS enable_tls=yes /* upper limit for TLS connections */ tls_max_connections=2048 #!endif ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ #!ifdef WITH_PSTN /* PSTN GW Routing * * - pstn.gw_ip: valid IP or hostname as string value, example: * pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" * * - by default is empty to avoid misrouting */ pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL /* VoiceMail Routing on offline, busy or no answer * * - by default Voicemail server IP is empty to avoid misrouting */ voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" loadmodule "dispatcher.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- dispatcher params ----- modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ # modparam("registrar", "max_contacts", 10) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- /* ip ban htable with autoexpire after 5 minutes */ modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains #route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN #route(PSTN); route(DISPATCH); # user location service route(LOCATION); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) { sl_reply_error(); } exit; } # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + if (starts_with("$rU", "00")) { strip(2); prefix("+"); } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif } *Regards:* Gaurav Kumar

1 0

Re: [SR-Users] disparcher making loop with call
by Gaurav Bmotra 22 Nov '19

22 Nov '19

hi thank you for reply i have check but same issue ,,, is the config file data plz check if you can help thank you ------------------------------------ #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ACCDB #!define WITH_NAT # # Kamailio (OpenSER) SIP Server v5.2 - default configuration script # - web: https://www.kamailio.org # - git: https://github.com/kamailio/kamailio # # Direct your questions about this file to: <sr-users(a)lists.kamailio.org> # # Refer to the Core CookBook at https://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Note: the comments can be: # - lines starting with #, but not the pre-processor directives, # which start with #!, like #!define, #!ifdef, #!endif, #!else, #!trydef, # #!subst, #!substdef, ... # - lines starting with // # - blocks enclosed in between /* */ # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To block 401 and 407 authentication replies execute: # - define WITH_BLOCK401407 # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes for each UDP socket * - value inherited by tcp_children and sctp_children when not set explicitely */ children=8 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes /* number of SIP routing processes for all TCP/TLS sockets */ # tcp_children=8 /* uncomment the next line to disable the auto discovery of local aliases * based on reverse DNS on IPs (default on) */ # auto_aliases=no /* add local domain aliases */ # alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to * bind on a specific interface/port/proto (default bind on all available) */ # listen=udp:10.0.0.10:5060 listen=udp:172.20.LOCAL_IP:5091 advertise 3.83.PUBLIC_IP:5091 /* life time of TCP connection when there is no traffic * - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 /* upper limit for TCP connections (it includes the TLS connections) */ tcp_max_connections=2048 #!ifdef WITH_TLS enable_tls=yes /* upper limit for TLS connections */ tls_max_connections=2048 #!endif ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ #!ifdef WITH_PSTN /* PSTN GW Routing * * - pstn.gw_ip: valid IP or hostname as string value, example: * pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" * * - by default is empty to avoid misrouting */ pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL /* VoiceMail Routing on offline, busy or no answer * * - by default Voicemail server IP is empty to avoid misrouting */ voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" loadmodule "dispatcher.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- dispatcher params ----- modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ # modparam("registrar", "max_contacts", 10) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- /* ip ban htable with autoexpire after 5 minutes */ modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains #route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN #route(PSTN); route(DISPATCH); # user location service route(LOCATION); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) { sl_reply_error(); } exit; } # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + if (starts_with("$rU", "00")) { strip(2); prefix("+"); } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif } ------------------------------------------------------------------------- On Fri, Nov 22, 2019 at 4:57 PM Gaurav Bmotra <saigauravmehra91(a)gmail.com> wrote: > hi thank you for reply > i have check but same issue > i have attached my config file ... can you check if there is any issue > > thank you > > On Fri, Nov 22, 2019 at 4:41 PM David Villasmil < > david.villasmil.work(a)gmail.com> wrote: > >> Hello, >> >> You’re only pasting the dispatcher route. >> >> You need to follow the script logic. An invite comes in, how does it know >> the call is coming from the client and has to be sent to freeSWITCH? >> Conversely, how does it know it’s coming from freeswitch? >> >> Probably using the permissions module with >> >> https://kamailio.org/docs/modules/3.1.x/modules_k/permissions.html#id2528655 >> >> You can check if the $si is the freeswitch, then send to clientB >> >> On Fri, 22 Nov 2019 at 11:40, Daniel-Constantin Mierla <miconda(a)gmail.com> >> wrote: >> >>> Hello, >>> >>> there is something wrong in your config file, failing to detect the >>> traffic from freeswitch. Load debugger module and enable cfgtrace logging, >>> then see what actions are executed in the configuration file, maybe you can >>> spot what goes wrong there. >>> >>> Otherwise, to get more help from the mailing lists, the full config is >>> needed. >>> >>> Cheers, >>> Daniel >>> On 22.11.19 09:51, Gaurav Bmotra wrote: >>> >>> hi >>> i m using kamailio 5.3 as register server and using dispatcher i m >>> dendign the calls to freeswitch .. >>> >>> but the issue is then freeswitch send back call to kamailio after >>> processing media ... it should go back to client but it again send call to >>> freeswitch ,, which create the loop and hangup the call with (Temporarily >>> Unavailable (code: 480)) >>> >>> >>> *what i want *client1------>Kamailio------>FS-----Kamailio----->Client2 >>> >>> *but I am getting this * >>> >>> client1------>Kamailio------>FS-----Kamailio----->FS----->kamailio----->FS >>> >>> >>> *Routing Logic *------------------------------- >>> if(!ds_is_from_list()) { >>> >>> route(DISPATCH); >>> } >>> >>> -------- >>> # Dispatch requests >>> >>> >>> >>> route[DISPATCH] { >>> >>> >>> >>> # round robin dispatching on gateways group '1' >>> >>> >>> >>> if(!ds_select_dst("1", "4")) { >>> >>> >>> send_reply("404", "No destination"); >>> >>> >>> exit; >>> >>> >>> >>> } >>> >>> >>> >>> xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: >>> $xavp(_dsdst_=>attrs))\n"); >>> >>> >>> t_on_failure("RTF_DISPATCH"); >>> >>> >>> >>> route(RELAY); >>> >>> >>> >>> exit; >>> >>> >>> >>> } >>> >>> >>> >>> >>> >>> >>> >>> # Try next destionations in failure route >>> >>> >>> >>> failure_route[RTF_DISPATCH] { >>> >>> >>> >>> xnotice("RTF_DISPATCH: $rm $rU [$ci]"); >>> >>> >>> >>> if (t_is_canceled()) { >>> >>> >>> exit; >>> >>> >>> >>> } >>> >>> >>> >>> # next DST - only for 500 or local timeout >>> >>> >>> if (t_check_status("(^5)") >>> >>> >>> or (t_branch_timeout() and !t_branch_replied())) >>> { >>> >>> xnotice("RTF_DISPATCH: WARNING $du is broken and marked >>> as inactive"); >>> >>> >>> # ds_mark_dst("ip"); >>> >>> >>> >>> if(ds_next_dst()) { >>> >>> >>> >>> #xdbg("--- SCRIPT: retrying to <$ru> via <$du> >>> (attrs: $xavp(_dsdst_=>attrs))\n"); >>> >>> >>> t_on_failure("RTF_DISPATCH"); >>> >>> >>> >>> route(RELAY); >>> >>> >>> >>> exit; >>> >>> >>> >>> } >>> >>> >>> >>> } >>> >>> >>> >>> } >>> >>> >>> -- >>> thank you >>> >>> >>> >>> >>> >>> >>> >>> *Regards:* >>> Gaurav Kumar >>> >>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >>> -- >>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >>> Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com >>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing List >>> sr-users(a)lists.kamailio.org >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >> -- >> Regards, >> >> David Villasmil >> email: david.villasmil.work(a)gmail.com >> phone: +34669448337 >> > > > -- > > > > > > > > > *Regards:* > Gaurav Kumar > > -- *Regards:* Gaurav Kumar

1 0

disparcher making loop with call
by Gaurav Bmotra 22 Nov '19

22 Nov '19

hi i m using kamailio 5.3 as register server and using dispatcher i m dendign the calls to freeswitch .. but the issue is then freeswitch send back call to kamailio after processing media ... it should go back to client but it again send call to freeswitch ,, which create the loop and hangup the call with (Temporarily Unavailable (code: 480)) *what i want *client1------>Kamailio------>FS-----Kamailio----->Client2 *but I am getting this * client1------>Kamailio------>FS-----Kamailio----->FS----->kamailio----->FS *Routing Logic*------------------------------- if(!ds_is_from_list()) { route(DISPATCH); } -------- # Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { xnotice("RTF_DISPATCH: $rm $rU [$ci]"); if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("(^5)") or (t_branch_timeout() and !t_branch_replied())) { xnotice("RTF_DISPATCH: WARNING $du is broken and marked as inactive"); # ds_mark_dst("ip"); if(ds_next_dst()) { #xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } -- thank you *Regards:* Gaurav Kumar

3 2

How to retrieve local socket that will be used to send a message from the branch route
by Patrick Wakano 22 Nov '19

22 Nov '19

Hello list, Hope you are all doing well! I am trying to figure out a way to retrieve the local socket the t_relay() has decided to use in a mhomed=1 env. Is there a way to do that? I couldn't find any variable other than the $snd ones that maybe have this info, but I am not using onsend_route anyway..... I am after this value because this is what I can use to figure out if I need to engage RTPEngine with parameters "external, internal", "external, external", "internal, external" or "internal, internal". In my case, my server is in between two networks, and calls can traverse or can be routed back to the same interface depending on what was decide by the LCR module. Signalling is fine, but engaging the RTPEngine must be done with the correct parameters to get the correct IPs in the SDP, other else I get one way audio in the call..... Any help is much appreciated! Thank you! Kind regards, Patrick Wakano

4 7

Adding SIP Gateway Endpoint in Kamailio
by Sujit Roy 21 Nov '19

21 Nov '19

Hello Here is my call flow scenario. SIP Gateway (A) Kamailio (K) Asterisk (AST) SIP Gateway (B) Now i want to send calls from A -> B by using Asterisk as media server. Kamailio shall be used to authenticate A and allow A to send calls to B. What are the configurations i need to make in Kamailio and Asterisk ? Thanks in advance. -- Regards =================== Sujit Roy

3 6

How to access different uris in a header
by Duarte Rocha 21 Nov '19

21 Nov '19

Greetings, In order to identify Redirecting Number and Original Called Number on a given request i look for the top( Redirecting Number) and bottom (Original Called) Diversion headers using hdr(Diversion)[0] and hdr(Diversion)[-1]. However, i have a client that sends me all the Diversions URIS in the same header. Does kamailio has a similar tool for accessing the different URIS in the header? Best Regards,

2 1

CentOS Repo for Kamailio 5.3 - Strange Python23 Dependency?
by Zach Nielsen 21 Nov '19

21 Nov '19

Hi all, Working with some of our Centos-based Kamailio boxes lately has produced a strange result when using the provided yum repos (download.opensuse.org). When trying to yum update the system or install some additional modules, we're getting feedback from yum indicating that it's expecting /usr/bin/python23 --> Finished Dependency Resolution Error: Package: kamailio-5.3.1-3.1.x86_64 (home_kamailio_v5.3.x-rpms) Requires: /usr/bin/python23 You could try using --skip-broken to work around the problem You could try running: rpm -Va --nofiles --nodigest Tried my best to do my due diligence here to make sure there was nothing up with our local env that was causing this, but this issue persists across multiple systems, including a new (no frills) one that we just build out. Can I get some confirmation if this is a "me" issue, or if maybe there's an issue with what made it into the repo? Also, just a general question... what's the typical way of you folks deploying Kamailio from a sysadmin perspective? Do you generally compile from source, or do any of you use the provided repositories to manage it? Thanks! Zach

5 6

kamailio Dispatcher error
by Gaurav Bmotra 20 Nov '19

20 Nov '19

hi i want to use dispatcher Module and following is the config.. --------------------------------------- loadmodule "dispatcher.so" -------------- modparam("dispatcher", "db_url", "mysql://kamailio:kamailiorw@localhost /kamailio") modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("dispatcher", "ds_hash_size", 2) -------------------------------------------------------------- but when i hit $ kamailio out put: ERROR: bad config file (4 errors) ------------------------------------------------------ logs : kamailio.service - Kamailio (OpenSER) - the Open Source SIP Server Loaded: loaded (/lib/systemd/system/kamailio.service; enabled; vendor preset: enabled) Active: failed (Result: exit-code) since Wed 2019-11-20 15:16:34 IST; 16s ago Process: 25747 ExecStart=/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f $CFGFILE -m $SHM_MEMORY -M $PKG_MEMORY -u $USER -g $GROUP (code=exited, status=255/EXCEPTION) Nov 20 15:16:34 gaurav-Latitude-E7240 systemd[1]: kamailio.service: Service RestartSec=100ms expired, scheduling restart. Nov 20 15:16:34 gaurav-Latitude-E7240 systemd[1]: kamailio.service: Scheduled restart job, restart counter is at 5. Nov 20 15:16:34 gaurav-Latitude-E7240 systemd[1]: Stopped Kamailio (OpenSER) - the Open Source SIP Server. Nov 20 15:16:34 gaurav-Latitude-E7240 systemd[1]: kamailio.service: Start request repeated too quickly. Nov 20 15:16:34 gaurav-Latitude-E7240 systemd[1]: kamailio.service: Failed with result 'exit-code'. Nov 20 15:16:34 gaurav-Latitude-E7240 systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. ------------------------------------------------------------- plz help me -- Thank you *Regards:* Gaurav Kumar

4 6

Dynamic routing and REGISTER
by Igor Olhovskiy 20 Nov '19

20 Nov '19

Hi! I’m trying to get `drouting` module working for REGISTER. But seems it’s not passing one of internal checks, cause in REGISTER R-URI does not contains username and I’m getting errors drouting [drouting.c:567]: build_ruri(): stripping 0 makes username <> null drouting [drouting.c:970]: do_routing(): failed to build ruri So, as I got, this module is mainly for calls (means INVITE) routing, rather than to use it like one more dispatcher, or it’s also a way to handle this part here? Thanks, Igor

3 5

Query regarding Kamailio Sip Server for IMS supporting VOLTE
by Shubhendu Das 20 Nov '19

20 Nov '19

Hi , I have a query regarding Kamailio open source SIP server. Does it have any release, where its supports the feature of IMS supporting VOLTE with real commercial mobile phones(IMSI based) ? If so, what is the release number ? Please also let me know about the supported mobile phones list. BR, Shubhendu -- _Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._

1 0

How to intercept an INVITE after (ts_append)
by Oleg Podguyko 20 Nov '19

20 Nov '19

Hello everybody! Kamailio works as a Registrar. Additionally, rtpengine is deployed next to it. Using rtpgngine module I proxy media stream via rtpengine(rtpengine_manage()). I change SDP. SiP2SIP calls work fine. Now I would like to have possibility to "wake up" sip-clients(at the android and ios phones) which are usually not on-line via push notification. Trying to use TSILO module. # User location service route[LOCATION] { lookup("location"); $var(rc) = $rc; if (is_method("INVITE")) { if (t_newtran()) { ts_store(); # if it a new transuction ,it will be store for some time. } } switch ($var(rc)) { case 1: route(RELAY); break; case -1: break; case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if (!save("location")) { sl_reply_error(); } ts_append("location", "$tu"); # if transuction for this sip-client is exist. The invite will be send. exit; } If the client is not registered at the kamailio during the call. Then such calls are stored through the "ts_store" function for some time. And if during this time the client send register, kamailio will be sent INVITE to it. But I have the problem. Such calls after "wake up"go through a special branch route that I don't know how to intercept. As a result, I can't change SDP(using rtpengite_manage commands) and rtp stream do not proxy via rtpengine… and call failed. /usr/sbin/kamailio[9957]: exec: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} *** cfgtrace:request_route=[REGISTRAR] c=[/etc/kamailio/kamailio.cfg] l=428 a=26 n=ts_append /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} tsilo [ts_append.c:72]: ts_append(): transaction 36343:92182831 found for 777777777, going to append branches /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} tm [t_lookup.c:1629]: t_lookup_ident_filter(): transaction found /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} registrar [lookup.c:255]: lookup_helper(): contact for [777777777] found by address /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} registrar [lookup.c:456]: lookup_helper(): instance is /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} tm [t_append_branches.c:77]: t_append_branches(): transaction 36343:92182831 in status 0 /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} tm [t_append_branches.c:99]: t_append_branches(): Call uvAzTGsEV9N2ReDjmWoo8A..: 0 (1) outgoing branches /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} tm [t_append_branches.c:126]: t_append_branches(): Current uri sip:77777777@95.165.146.181:63618;transport=TCP;rinstance=8b1865b95767b7ba /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} <core> [core/msg_translator.c:2933]: create_via_hf(): id added: <;i=1>, rcv proto=2 /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} tm [t_append_branches.c:149]: t_append_branches(): added branch [sip:777777777@95.165.146.181:63618;transport=TCP;rinstance=8b1865b95767b7ba] with ruid [uloc-5dd2faa0-26e5-1] /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} tm [t_append_branches.c:163]: t_append_branches(): Call uvAzTGsEV9N2ReDjmWoo8A..: 0 (0) outgoing branches after clear_branches() /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} <core> [core/tcp_main.c:2238]: tcpconn_send_put(): send from reader (9957 (22)), reusing fd /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} <core> [core/tcp_main.c:2473]: tcpconn_do_send(): sending... -- Oleg Podguyko

2 2

TLS v 1.3 failed call
by Andrew Chen 19 Nov '19

19 Nov '19

Hi guys, I was wondering if someone can help decipher what these few lines mean? Nov 19 20:12:50 sjomainkama51 /usr/sbin/kamailio[21501]: DEBUG: tls [tls_server.c:431]: tls_accept(): tls_accept: new connection from 10.94.98.18:51698 using TLSv1.3 TLS_AES_256_GCM_SHA384 256 Nov 19 20:12:50 sjomainkama51 /usr/sbin/kamailio[21501]: DEBUG: tls [tls_server.c:434]: tls_accept(): tls_accept: local socket: 206.81.191.45:443 Nov 19 20:12:50 sjomainkama51 /usr/sbin/kamailio[21501]: DEBUG: tls [tls_server.c:445]: tls_accept(): tls_accept: client did not present a certificate Nov 19 20:12:50 sjomainkama51 /usr/sbin/kamailio[21501]: DEBUG: tls [tls_server.c:1189]: tls_read_f(): Reading on a renegotiation of connection (n:1652) (0) Nov 19 20:12:50 sjomainkama51 /usr/sbin/kamailio[21501]: DEBUG: <core> [core/tcp_read.c:1527]: tcp_read_req(): EOF Nov 19 20:12:50 sjomainkama51 /usr/sbin/kamailio[21501]: DEBUG: <core> [core/io_wait.h:602]: io_watch_del(): DBG: io_watch_del (0x56367f8b8f80, 12, -1, 0x10) fd_no=2 called Nov 19 20:12:50 sjomainkama51 /usr/sbin/kamailio[21501]: DEBUG: <core> [core/tcp_read.c:1680]: release_tcpconn(): releasing con 0x7fcfc20c57b0, state -1, fd=12, id=23 ([10.94.98.18]:51698 -> [10.94.98.18]:443) Nov 19 20:12:50 sjomainkama51 /usr/sbin/kamailio[21501]: DEBUG: <core> [core/tcp_read.c:1684]: release_tcpconn(): extra_data 0x7fcfc2089f88 Nov 19 20:12:50 sjomainkama51 /usr/sbin/kamailio[21517]: DEBUG: <core> [core/tcp_main.c:3320]: handle_tcp_child(): reader response= 7fcfc20c57b0, -1 from 16 Nov 19 20:12:50 sjomainkama51 /usr/sbin/kamailio[21517]: DEBUG: tls [tls_server.c:683]: tls_h_close(): Closing SSL connection 0x7fcfc2089f88 Bottom line is kamailio closed the connection and I have method setting to this: root@sjomainkama51:/etc/kamailio # grep method tls.cfg method = TLSv1.1+ Supposedly this should work? -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 2

Fwd: Websocket immediately disconnecting
by Tom D 19 Nov '19

19 Nov '19

Gentlemen, Thanks for your feedback. Yes, I've doubled the RAM last night (between my posts) and I am monitoring the behavior. It *feels* that it is definitely working better. I haven't enabled memjoin yet - are there any repercussions that should be expected for enabling memjoin=yes? Thanks for your guidance. Tom

3 2

Websocket immediately disconnecting
by Tom D 19 Nov '19

19 Nov '19

Hi David, Thanks for your reply. After increasing the verbosity of the debugging, I found the following entries. So am I understanding the log correctly: the websocket module is trying to get some free memory (from Kamilio's general pool?). It doesn't receive any yet proceeds with parsing the response (I get this same parsing error even when it's working) and ultimately, it doesn't register the websocket? I appreciate your (or anyone's) thoughts. Regards, Tom Nov 19 02:44:44 MY-HOSTNAME /usr/sbin/kamailio[14928]: CRITICAL: tls [tls_domain.c:737]: sr_ssl_ctx_info_callback(): SSL handshake started Nov 19 02:44:44 MY-HOSTNAME /usr/sbin/kamailio[14928]: CRITICAL: tls [tls_domain.c:749]: sr_ssl_ctx_info_callback(): SSL handshake done Nov 19 02:44:44 MY-HOSTNAME /usr/sbin/kamailio[14928]: CRITICAL: tls [tls_domain.c:753]: sr_ssl_ctx_info_callback(): SSL disable renegotiation Nov 19 02:44:44 MY-HOSTNAME /usr/sbin/kamailio[14928]: CRITICAL: tls [tls_server.c:431]: tls_accept(): tls_accept: new connection from HI.DD.E.N:63356 using TLSv1/SSLv3 AES256-GCM-SHA384 256 Nov 19 02:44:44 MY-HOSTNAME /usr/sbin/kamailio[14928]: CRITICAL: tls [tls_server.c:434]: tls_accept(): tls_accept: local socket: HI.DD.E.N:8089 Nov 19 02:44:44 MY-HOSTNAME /usr/sbin/kamailio[14928]: CRITICAL: tls [tls_server.c:445]: tls_accept(): tls_accept: client did not present a certificate Nov 19 02:44:45 MY-HOSTNAME /usr/sbin/kamailio[14928]: CRITICAL: <script>: 857:HTTP Request Received Nov 19 02:44:45 MY-HOSTNAME /usr/sbin/kamailio[14928]: ERROR: <core> [core/mem/q_malloc.c:291]: qm_find_free(): qm_find_free(0x7f8b8cdcd000, 65728); Free fragment not found! Nov 19 02:44:45 MY-HOSTNAME /usr/sbin/kamailio[14928]: ERROR: <core> [core/mem/q_malloc.c:425]: qm_malloc(): qm_malloc(0x7f8b8cdcd000, 65728) called from websocket: ws_conn.c: wsconn_add(197), module: websocket; Free fragment not found! Nov 19 02:44:45 MY-HOSTNAME /usr/sbin/kamailio[14928]: ERROR: websocket [ws_conn.c:199]: wsconn_add(): allocating shared memory Nov 19 02:44:45 MY-HOSTNAME /usr/sbin/kamailio[14928]: ERROR: <core> [core/parser/parse_fline.c:262]: parse_first_line(): parse_first_line: bad message (offset: 22) Nov 19 02:44:45 MY-HOSTNAME /usr/sbin/kamailio[14928]: ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<HTTP/1.1 101 Switching Protocols#015#012Sia: SIP/2.0/TLS HI.DD.E.N:63356#015#012Access-Control-Allow-Origin: https://www.doubango.org#015#012Sec-WebSocket-Protocol: sip#015#012Upgrade: websocket#015#012Connection: upgrade#015#012Sec-WebSocket-Accept: taS0s3Z7t4tOhklSbJDLGmSSYsY=#015#012Server: kamailio (5.2.5 (x86_64/linux))#015#012Content-Length: 0#015#012#015#012> Nov 19 02:44:45 MY-HOSTNAME /usr/sbin/kamailio[14928]: ERROR: websocket [ws_frame.c:583]: ws_frame_receive(): WebSocket connection not found

3 3

Asterisk text messages not delivered to SIP client
by Adesh Pandey 19 Nov '19

19 Nov '19

Hi Guys, I have been using Kamailio as dispatcher but I have been facing a weird situation that when asterisk SendText command says that message was delivered successfully but webrtc sip client did not receive the message. My deployment has Kamailio as registrar where sip users are registered and when they make any outgoing call, the call is proxied by RTPEngine and then dispatched to Asterisk boxes. Asterisk boxes send some text to the client using SendText but the message is not received by the SIP client. My config as follows: https://gist.github.com/adeshpandey/2c338db7f5d992267f7415de0325e0b4

1 0

Kamailio meeting Istanbul and Cairo
by Henning Westerholt 19 Nov '19

19 Nov '19

Hello, I will be in about 2 weeks in Istanbul and in Cairo a few days later. I am thinking of organizing a Kamailio/SIP/VoIP social networking event (perhaps a dinner or a pub meeting). Possible dates are the 19th- 20th November in Istanbul and the 23th - 24th November in Cairo. Are people from the community interested in joining such event? If yes, write me your preferred date -- based on feedback in the next days, it will be decided whether it is going to happen and when. Best regards, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

2 2

Re: [SR-Users] call disconnected in 45 min
by Liu, Charles 18 Nov '19

18 Nov '19

Hi All, I have install kamailio IMS using the example config files for P/S/I-cscf with version 5.2... call randomly disconnected between 15 to 45 minutes. Wireshark log show scscf sent BYE to the caller and callee. Any idea what maybe the problem? Scscf debug output ......... ............ 20(25) INFO: ims_registrar_scscf [cxdx_avp.c:138]: cxdx_get_avp(): cxdx_get_experimental_result_code: Failed finding avp 20(25) INFO: ims_registrar_scscf [cxdx_sar.c:84]: create_return_code(): created AVP successfully : [saa_return_code] - [1] 13(18) DEBUG: ims_dialog [dlg_handlers.c:1923]: print_all_dlgs(): ********************13(18) DEBUG: ims_dialog [dlg_handlers.c:1924]: print_all_dlgs(): printing 4096 dialogs 13(18) DEBUG: ims_dialog [dlg_handlers.c:1862]: internal_print_all_dlg(): Trying to get lock for printing 13(18) DEBUG: ims_dialog [dlg_handlers.c:1868]: internal_print_all_dlg(): ----------------------------13(18) DEBUG: ims_dialog [dlg_handlers.c:1869]: internal_print_all_dlg(): Dialog h_entry:h_id = [3396 : 2798] 13(18) DEBUG: ims_dialog [dlg_handlers.c:1870]: internal_print_all_dlg(): Dialog age: 2749 13(18) DEBUG: ims_dialog [dlg_handlers.c:1871]: internal_print_all_dlg(): Dialog call-id: ATokIPEjouNpuwSo6MpHUw.. 13(18) DEBUG: ims_dialog [dlg_handlers.c:1872]: internal_print_all_dlg(): Dialog state: 4 13(18) DEBUG: ims_dialog [dlg_handlers.c:1873]: internal_print_all_dlg(): Dialog ref counter: 2 13(18) DEBUG: ims_dialog [dlg_handlers.c:1874]: internal_print_all_dlg(): Dialog did: 13(18) DEBUG: ims_dialog [dlg_handlers.c:1875]: internal_print_all_dlg(): Dialog from_tag: 8975d611 13(18) DEBUG: ims_dialog [dlg_handlers.c:1876]: internal_print_all_dlg(): Dialog from_uri: sip:bob@ims.mnc001.mcc001.3gppnetwork.org;transport=UDP 13(18) DEBUG: ims_dialog [dlg_handlers.c:1877]: internal_print_all_dlg(): Dialog caller contact: sip:bob@192.168.164.23:39180;transport=UDP 13(18) DEBUG: ims_dialog [dlg_handlers.c:1878]: internal_print_all_dlg(): Dialog first request cseq: 1 13(18) DEBUG: ims_dialog [dlg_handlers.c:1879]: internal_print_all_dlg(): Dialog caller route set: <sip:mo@10.1.1.1;lr=on;ftag=8975d611;rm=8> 13(18) DEBUG: ims_dialog [dlg_handlers.c:1880]: internal_print_all_dlg(): Dialog lifetime: 43200 13(18) DEBUG: ims_dialog [dlg_handlers.c:1881]: internal_print_all_dlg(): Dialog bind_address: udp:172.26.1.60:6060 13(18) DEBUG: ims_dialog [dlg_handlers.c:1882]: internal_print_all_dlg(): Dialog-creating transaction pointer [0x7f8bb2a51858] 13(18) DEBUG: ims_dialog [dlg_handlers.c:1888]: internal_print_all_dlg(): ----------13(18) DEBUG: ims_dialog [dlg_handlers.c:1889]: internal_print_all_dlg(): Dialog out h_entry:h_id = [3296 : 0] 13(18) DEBUG: ims_dialog [dlg_handlers.c:1890]: internal_print_all_dlg(): Dialog out did: 13(18) DEBUG: ims_dialog [dlg_handlers.c:1891]: internal_print_all_dlg(): Dialog out to_tag: c9eb5e55 13(18) DEBUG: ims_dialog [dlg_handlers.c:1892]: internal_print_all_dlg(): Dialog out caller cseq: 1 13(18) DEBUG: ims_dialog [dlg_handlers.c:1893]: internal_print_all_dlg(): Dialog out callee cseq: 1 13(18) DEBUG: ims_dialog [dlg_handlers.c:1894]: internal_print_all_dlg(): Dialog out callee contact: sip:alice@192.168.164.84:49153;alias=192.168.164.84~49153~1 13(18) DEBUG: ims_dialog [dlg_handlers.c:1895]: internal_print_all_dlg(): Dialog out callee route set: <sip:mt@172.26.1.60:6060;lr;ftag=8975d611;did=44d.eea>,<sip:mt@10.1.1.1;lr;ftag=8975d611;rm=7> 13(18) DEBUG: ims_dialog [dlg_handlers.c:1896]: internal_print_all_dlg(): Dialog out branch: z9hG4bKb7d5.6f11b9b34a8a493a3a1b7949831e2f47.0 13(18) DEBUG: ims_dialog [dlg_handlers.c:1897]: internal_print_all_dlg(): Dialog out state (deleted): 0 13(18) DEBUG: ims_dialog [dlg_handlers.c:1899]: internal_print_all_dlg(): ----------13(18) DEBUG: ims_dialog [dlg_handlers.c:1903]: internal_print_all_dlg(): Releasing lock for dlgout 13(18) DEBUG: ims_dialog [dlg_handlers.c:1906]: internal_print_all_dlg(): ----------------------------13(18) DEBUG: ims_dialog [dlg_handlers.c:1934]: print_all_dlgs(): ********************13(18) DEBUG: ims_auth [authorize.c:187]: reg_await_timer(): Looking for expired/useless at 44469727 13(18) DEBUG: ims_auth [authorize.c:192]: reg_await_timer(): Slot 121 <alice(a)ims.mnc001.mcc001.3gppnetwork.org<mailto:alice@ims.mnc001.mcc001.3gppnetwork.org>> 13(18) DEBUG: ims_auth [authorize.c:197]: reg_await_timer(): .. AV 1 - 3 Exp 44469993 0x7f8bb2a5c988 13(18) DEBUG: ims_auth [authorize.c:192]: reg_await_timer(): Slot 914 <bob(a)ims.mnc001.mcc001.3gppnetwork.org<mailto:bob@ims.mnc001.mcc001.3gppnetwork.org>> 13(18) DEBUG: ims_auth [authorize.c:197]: reg_await_timer(): .. AV 1 - 3 Exp 44470022 0x7f8bb2a54e88 13(18) DEBUG: ims_auth [authorize.c:197]: reg_await_timer(): .. AV 1 - 2 Exp 44469974 0x7f8bb2a506a8 13(18) DEBUG: ims_auth [authorize.c:203]: reg_await_timer(): ... dropping av 1 - 2 13(18) DEBUG: ims_auth [authorize.c:232]: reg_await_timer(): [DONE] Looking for expired/useless at 44469727 13(18) DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7f8bb2a551f8 with 1 -> 3 13(18) DEBUG: ims_dialog [dlg_hash.c:889]: lookup_dlg(): dialog id=2798 found on entry 3396 13(18) DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7f8bb2a551f8 with 1 -> 2 13(18) DEBUG: ims_dialog [dlg_req_within.c:379]: dlg_terminate(): terminating confirmed dialog 13(18) DEBUG: ims_dialog [dlg_req_within.c:272]: send_bye(): sending BYE to caller 13(18) DEBUG: ims_dialog [dlg_hash.c:1047]: ref_dlg(): ref dlg 0x7f8bb2a551f8 with 1 -> 3 13(18) DEBUG: ims_dialog [dlg_req_within.c:289]: send_bye(): BYE sent to caller 13(18) DEBUG: ims_dialog [dlg_req_within.c:272]: send_bye(): sending BYE to callee 13(18) DEBUG: ims_dialog [dlg_hash.c:1047]: ref_dlg(): ref dlg 0x7f8bb2a551f8 with 1 -> 4 13(18) DEBUG: ims_dialog [dlg_req_within.c:289]: send_bye(): BYE sent to callee 1(6) ERROR: <script>: BYE (sip:bob@ims.mnc001.mcc001.3gppnetwork.org;transport=UDP (172.26.1.60:6060) to sip:alice@ims.mnc001.mcc001.3gppnetwork.org, ATokIPEjouNpuwSo6MpHUw..) 1(6) DEBUG: ims_dialog [dlg_handlers.c:844]: dlg_onroute(): route param is '44d.eea' (len=7) 1(6) DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7f8bb2a551f8 with 1 -> 5 1(6) DEBUG: ims_dialog [dlg_hash.c:889]: lookup_dlg(): dialog id=2798 found on entry 3396 1(6) DEBUG: ims_dialog [dlg_hash.h:459]: match_dialog(): searching dlg_outs 1(6) DEBUG: ims_dialog [dlg_hash.h:496]: match_dialog(): p: 0x7f8bb2a68c48 ft[8975d611] tt [c9eb5e55] 1(6) DEBUG: ims_dialog [dlg_hash.h:520]: match_dialog(): dout: tt[c9eb5e55] 1(6) DEBUG: ims_dialog [dlg_profile.c:419]: set_current_dialog(): setting current dialog [3396:2798] 1(6) DEBUG: ims_dialog [dlg_hash.c:1293]: next_state_dlg(): dialog 0x7f8bb2a551f8 changed from state 4 to state 6, due event 7 CONFIDENTIALITY NOTICE: This email and any attachments are for the sole use of the intended recipient and may contain material that is proprietary, confidential, privileged or otherwise legally protected or restricted under applicable government laws. Any review, disclosure, distributing or other use without expressed permission of the sender is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies without reading, printing, or saving.

1 0

Re: [SR-Users] Saving REGISTER status in kamailio database while REGSITER packets are forwarded.
by Voip support 18 Nov '19

18 Nov '19

I figured this finally out and all works for me very well. In usrloc dbmode i set value of 3 (its saving immediately changes into DB). In request route for REGISTER method i set t_on_reply("MANAGE_REPLY_REGISTER" and for REGISTER with expires set to 0 i set t_on_reply("MANAGE_REPLY_UNREGISTER") (used pseudovariable $expires(max)). I stored Contact header in $avp(contact) for unREGISTER request for further use when calling unregister(). Then in onreply_route for MANAGE_REPLY_REGISTER i call save lcoation and in MANAGE_REPLY_UNREGISTER for reply code 200 i call unregister function with the stored contact (i had to remove < > and display name using re.subst). So far it looks to work perfect. pon., 18 lis 2019 o 13:38 Voip support <voipexpert0(a)gmail.com> napisał(a): > > Thanks again for all replies... I need to try it out - did not have time > for it yet. > > I think i will go for the onreply_route answer and try to save location in > this place. > > pon., 18 lis 2019 o 06:23 Joel Serrano <joel(a)textplus.com> napisał(a): > >> Yes sir! Works perfectly! >> >> To give you a quick example: >> >> 1) Add in the appropriate place for the REGISTER request: >> t_on_reply("MANAGE_REG_REPLY"); >> 2) Create a new onreply route: >> >> # Manage registration incoming replies >> onreply_route[MANAGE_REG_REPLY] { >> if(status=~"2[0-9][0-9]") { >> if (save("location", "0x04")) { >> xlog("L_NOTICE", "[usrloc] saved user location\n"); >> } >> } >> } >> >> >> Note1: In my setup, I needed to strictly simulate asterisk (chan_sip) >> registrations, so only one AoR was allowed, thus the flags 0x04, if you are >> using PJSIP you might not need it. >> Note2: With this example code, a "unregister" (REGISTER with expires=0) >> will not remove the AoR from Kamailio, to achieve that I check >> independently if the header Expires exists and is 0, or if the contact >> param Expires exists and is 0, and in that case I use the unregister() >> function instead of the save(). >> >> Hope it helps! >> Joel. >> >> >> >> On Sun, Nov 17, 2019 at 5:42 PM Ovidiu Sas <osas(a)voipembedded.com> wrote: >> >>> Did it work calling save() on the reply route? >>> >>> -ovidiu >>> >>> On Thu, Nov 14, 2019 at 6:20 PM Voip support <voipexpert0(a)gmail.com> >>> wrote: >>> > >>> > I am using kamailio 5.3. >>> > >>> > Ovidiu Sas, i will try this tomorrow today is bit late. Maybe it could >>> work. >>> > >>> > David Villasmil , i know i can use SQLOPS but i was thinking of easier >>> method of doing this. >>> > By successful REGISTER i mean save only REGISTER attempt that >>> authenticated correctly at SIP server and in 200 OK a binding was returned. >>> > Same if the binding is later removed i would like remove it from >>> database. I need this because when i run REGISTER over kamailio to >>> destination SIP server i see all register from same IP. >>> > I need to be able to check what was the real source IP of client >>> REGISTER. >>> > >>> > Regards, >>> > Tomasz >>> > >>> > >>> > >>> > pt., 15 lis 2019 o 00:16 Mike from ApoloCore <mike(a)apolocore.com> >>> napisał(a): >>> >> >>> >> I am using kamailio 5.3. >>> >> >>> >> Ovidiu Sas, i will try this tomorrow today is bit late. Maybe it >>> could work. >>> >> >>> >> David Villasmil , i know i can use SQLOPS but i was thinking of >>> easier method of doing this. >>> >> By successful REGISTER i mean save only REGISTER attempt that >>> authenticated correctly at SIP server and in 200 OK a binding was returned. >>> >> Same if the binding is later removed i would like remove it from >>> database. I need this because when i run REGISTER over kamailio to >>> destination SIP server i see all register from same IP. >>> >> I need to be able to check what was the real source IP of client >>> REGISTER. >>> >> >>> >> Regards, >>> >> Tomasz >>> >> >>> >> >>> >> >>> >> >>> >> czw., 14 lis 2019 o 23:33 Ovidiu Sas <osas(a)voipembedded.com> >>> napisał(a): >>> >>> >>> >>> Try to use save() when you process the 200 ok reply for REGISTER in >>> >>> the reply route. >>> >>> >>> >>> Regards, >>> >>> Ovidiu Sas >>> >>> >>> >>> On Thu, Nov 14, 2019 at 3:40 PM Voip support <voipexpert0(a)gmail.com> >>> wrote: >>> >>> > >>> >>> > I setup kamailio to forward REGISTER packets using dispatcher to >>> two servers main and backup. >>> >>> > >>> >>> > The routing logic is quite simple i use the dispatcher probing >>> mechanism to detect if my main dispatcher set id (one server) is active or >>> inactive once it detect a failure all packets are sent to secondary SIP >>> server to achieve High Availability. >>> >>> > >>> >>> > I am looking for module or a way to save into kamailio database >>> client information such as IP address, register username (from FROM >>> header), digest username (authorization username), call-id for successful >>> registration. >>> >>> > >>> >>> > I tried to use registrar module but once i used function >>> save("location") the client was registered on kamailio and packets was not >>> forwarded to main/secondary SIP server (like dispatcher stopped working, >>> the destination did not override). >>> >>> > >>> >>> > REGISTER is sent to SIP URI of kamailio. >>> >>> > >>> >>> > Please let me know if you have any idea how to make this data >>> update in database (insert an entry on fresh successfull REGISTER, delete >>> on expired or removed binding). >>> >>> > >>> >>> > Best regards, >>> >>> > Tomasz >>> >>> > >>> >>> > _______________________________________________ >>> >>> > Kamailio (SER) - Users Mailing List >>> >>> > sr-users(a)lists.kamailio.org >>> >>> > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >>> >>> >>> >>> >>> >>> -- >>> >>> VoIP Embedded, Inc. >>> >>> http://www.voipembedded.com >>> >>> >>> >>> _______________________________________________ >>> >>> Kamailio (SER) - Users Mailing List >>> >>> sr-users(a)lists.kamailio.org >>> >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> > >>> > _______________________________________________ >>> > Kamailio (SER) - Users Mailing List >>> > sr-users(a)lists.kamailio.org >>> > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >>> >>> -- >>> VoIP Embedded, Inc. >>> http://www.voipembedded.com >>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing List >>> sr-users(a)lists.kamailio.org >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >>

1 0

Websocket immediately disconnecting
by Tom D 18 Nov '19

18 Nov '19

Hi Everyone, We're running Kamailio 5.2.5 with XHTTP, Websocket and TLS extensions - for the purpose of being able to provide WebRTC capability. The settings for the HTTP request are quite default as set out below. The SIP client we're using is SIP5ML in different web browsers. After a successful start of Kamailio, it is possible to use SIP5ML and its features - i.e. it REGISTER's properly, calls can be made, etc. After about a few hours of running, Kamailio refuses to interact and abruptly closes the TCP connection. The only solution (discovered so far) is to restart Kamailio, which is not feasible on a server that is serving production needs. (On our development server, where the load is only a fraction - this problem does not occur). When the problems does start to occur, in Wireshark I see: 1) Normal TLS negotiation and certificate exchange; followed by 2) SIP5ML initiating the initial HTTP GET 3) Kamailio responds to the upgrade with a 101/Switch Protocols. (4- SIP5ML outputs a SIP REGISTER on a defunct socket as its was closed) Immediately after the 101/Switching Protocol message is received, Kamailio closes down the TCP connection with a FIN message. Nothing in the message or Kamailio log files help to identify what module is responsible for this. Would anyone have any advise where to look further? BTW, I don't see the TCP connection being Reset in the exchanges - so any advise will be appreciated. Tom event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != 8089) { xlogl("L_CRIT", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden - 0x24", "", ""); exit; } xlogl("L_CRIT", "HTTP Request Received\n"); if ($hdr(Upgrade)=~"websocket" && $hdr(Connection)=~"Upgrade" && $rm=~"GET") { # Validate Host - make sure the client is using the correct # alias for WebSockets if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { xlogl("L_CRIT", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden - 0x25", "", ""); exit; } # ws_handle_handshake() exits (no further configuration file # processing of the request) when complete. if (ws_handle_handshake()) { # Optional... cache some information about the # successful connection exit; } } xhttp_reply("404", "Resource Not Found - 0x27", "", ""); } event_route[websocket:closed] { xlogl("L_CRIT", "WebSocket connection from $si:$sp has closed\n"); }

2 1

Kamailio stop responding to AWS R53 health check
by Andrew Chen 18 Nov '19

18 Nov '19

Hi all, R53 reported healthcheck failure on tcp port 5060. Looking at the system, all the ports that kamailio listens on can't be monitored remotely. To test this theory, I can telnet successfully to consul port. I ran tcpdump and see no responses returned back to R53 monitoring hosts: tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on ens5, link-type EN10MB (Ethernet), capture size 262144 bytes 19:11:49.368378 IP ec2-54-241-32-75.us-west-1.compute.amazonaws.com.10072 > sjoprodkama51.fuzemeeting.com.sip: Flags [S], seq 1252984048, win 29200, options [mss 1460,sackOK,TS val 3770272726 ecr 0,nop,wscale 7], length 0 19:11:49.558877 IP ec2-107-23-255-43.compute-1.amazonaws.com.60794 > sjoprodkama51.fuzemeeting.com.sip: Flags [S], seq 3484910496, win 29200, options [mss 1460,sackOK,TS val 1780607162 ecr 0,nop,wscale 7], length 0 19:11:50.572870 IP ec2-107-23-255-43.compute-1.amazonaws.com.60794 > sjoprodkama51.fuzemeeting.com.sip: Flags [S], seq 3484910496, win 29200, options [mss 1460,sackOK,TS val 1780608175 ecr 0,nop,wscale 7], length 0 Fixed the issue after restarting kamailio process. Given this system is support production traffic, we have minimized the logging on it. The events that we thought may lead to this is as follows: Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd960380 r: 0x7f33fd960400 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda01dc0 r: 0x7f33fda01e40 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20310]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20310]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdd829e8 r: 0x7f33fdd82a68 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdb70738 r: 0x7f33fdb707b8 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20310]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20310]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20310]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdd829e8 r: 0x7f33fdd82a68 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda01dc0 r: 0x7f33fda01e40 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdaec8f0 r: 0x7f33fdaec970 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd93e6c0 r: 0x7f33fd93e740 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdaec8f0 r: 0x7f33fdaec970 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20307]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20307]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd960380 r: 0x7f33fd960400 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd960380 r: 0x7f33fd960400 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20307]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20307]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdaec8f0 r: 0x7f33fdaec970 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20307]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20307]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdd829e8 r: 0x7f33fdd82a68 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20309]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20309]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20309]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20309]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda69958 r: 0x7f33fda699d8 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20309]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20309]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdd829e8 r: 0x7f33fdd82a68 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda25fd8 r: 0x7f33fda26058 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20311]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20311]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda69958 r: 0x7f33fda699d8 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20309]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20309]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdd829e8 r: 0x7f33fdd82a68 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20311]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20311]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda25fd8 r: 0x7f33fda26058 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20311]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20311]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda25fd8 r: 0x7f33fda26058 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20311]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20311]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20314]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20314]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda25fd8 r: 0x7f33fda26058 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20312]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20312]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20312]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20312]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda25fd8 r: 0x7f33fda26058 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20312]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20312]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdbd5d78 r: 0x7f33fdbd5df8 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20314]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20314]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdb822d0 r: 0x7f33fdb82350 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20314]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20314]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdb822d0 r: 0x7f33fdb82350 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda01dc0 r: 0x7f33fda01e40 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdaec8f0 r: 0x7f33fdaec970 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda25fd8 r: 0x7f33fda26058 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdacac30 r: 0x7f33fdacacb0 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20314]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20314]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdaec8f0 r: 0x7f33fdaec970 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20317]: BUG: tls [tls_server.c:1303]: tls_read_f(): IO error (0) Success Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20317]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdacac30 r: 0x7f33fdacacb0 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdbb40b8 r: 0x7f33fdbb4138 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda01dc0 r: 0x7f33fda01e40 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdaa8f70 r: 0x7f33fdaa8ff0 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdb70738 r: 0x7f33fdb707b8 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdc7f490 r: 0x7f33fdc7f510 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20317]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20317]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdb70738 r: 0x7f33fdb707b8 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20314]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20314]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdbb40b8 r: 0x7f33fdbb4138 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdacac30 r: 0x7f33fdacacb0 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd960380 r: 0x7f33fd960400 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda01dc0 r: 0x7f33fda01e40 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda01dc0 r: 0x7f33fda01e40 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd960380 r: 0x7f33fd960400 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdacac30 r: 0x7f33fdacacb0 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20307]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdaa8f70 r: 0x7f33fdaa8ff0 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20307]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd93e6c0 r: 0x7f33fd93e740 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdb70738 r: 0x7f33fdb707b8 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20315]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20315]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdb822d0 r: 0x7f33fdb82350 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20316]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20316]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda25fd8 r: 0x7f33fda26058 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdc196f8 r: 0x7f33fdc19778 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Any thoughts on what would cause this issue? Thanks -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

4 14

Re: [SR-Users] Parsing Contact Port Field
by Michael Iedema 18 Nov '19

18 Nov '19

> Oops. No, it’s not a bug, it’s just that the URI isn’t the only value present in $ct. > > Try $(ct{nameaddr.uri}{uri.host}) Yup, that fixed it. Thanks again! I have another question about how I’m applying these to some routing logic but I’ll start a new thread for that in a bit.

1 0

http_async_client and HTTP2
by Igor Olhovskiy 18 Nov '19

18 Nov '19

HI! Seems to be, http_async client is not compatible with http2 servers. I'm having like kamailio_1 | 15(20) INFO: http_async_client [http_multi.c:238]: debug_cb(): [cURL] Connection #0 to host relay-api.pbxservice.com left intact kamailio_1 | 15(20) ERROR: <core> [core/parser/parse_fline.c:264]: parse_first_line(): parse_first_line: bad message (offset: 11) kamailio_1 | 15(20) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<HTTP/2 404 kamailio_1 | date: Thu, 14 Nov 2019 18:31:34 GMT .... kamailio_1 | 15(20) ERROR: <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [(null)] kamailio_1 | 15(20) ERROR: <core> [core/parser/msg_parser.c:1274]: get_body(): failed to parse to end of headers But plain http_client worked good (not async for sure) kamailio 5.3.0 -- Best regards, Igor

3 3

Problem with URI transformations
by Duarte Rocha 18 Nov '19

18 Nov '19

Greetings, I'm trying to parse some URI info on Kamailio but i'm having some troubles the URI transformations. Here i have two examples of URIS from which i'm trying to extract username(with user parameters included) and URI Parameters : - sip:123456789;test=param@1.1.1.1 - sip:123456789;test=param@1.1.1.1;user=phone In order to test this i made the following code : $var(URI) = "sip:123456789;test=param@1.1.1.1"; xerr("DEBUG 1 : URI $var(URI)"); $var(User) = $(var(URI){uri.user}); $var(Parameters) = $(var(URI){uri.params}); xerr("DEBUG 1 : User = $var(User)"); xerr("DEBUG 1 : Parameters = $var(Parameters)"); $var(URI) = "sip:123456789;test=param@1.1.1.1;user=phone"; xerr("DEBUG 2 : URI $var(URI)"); ; $var(User) = $(var(URI){uri.user}); $var(Parameters) = $(var(URI){uri.params}); xerr("DEBUG 2 : User = $var(User)"); xerr("DEBUG 2 : Parameters = $var(Parameters)"); The result is the following : DEBUG 1 : URI sip:123456789;test=param@1.1.1.1 DEBUG 1 : User = 123456789;test=param DEBUG 1 : Parameters = DEBUG 2 : URI sip:123456789;test=param@1.1.1.1;user=phone DEBUG 2 : User = 123456789 DEBUG 2 : Parameters = user=phone If the URI has parameters, the User parameters won't be included in the User part. Is this a bug or is it working as intended? I'm running Kamailio 5.2.0 . Thank you for your time. Best Regards, Duarte Rocha

3 6

Parsing Contact Port Field
by Michael Iedema 18 Nov '19

18 Nov '19

Hello everyone, I have a potentially silly beginners question: how can I parse the contact port field and act on it in my routing logic? I know that the $ct variable contains the entire contact header and I can print it in an xlog() call. However, I’d like to do something like the following in my routing logic: . . . SIP CONTENT Contact: <sip:someaccount@192.168.86.104:5062>;expires=1800 . . . CONFIG LOGIC #!define PORT_SERVICE1 5061 #!define PORT_SERVICE2 5062 route { if ( src_port == PORT_SERVICE1 ) { xlog(“L_INFO”, “Received $rm from SERVICE1\n”); } else if ( src_port == PORT_SERVICE2 ) { xlog(“L_INFO”, “Received $rm from SERVICE2\n”); } } I realize that src_port is not the correct value to compare against. I want to compare against the originating contact’s port value. How can I extract the port field from $ct? Many thanks in advance and apologies for the beginners question. I’ve googled for what I think I’m trying to do without any real results on functions or tokenizers, etc. I’m using Kamailio 5.3.0. Regards, -Michael

3 3

Re: [SR-Users] Saving REGISTER status in kamailio database while REGSITER packets are forwarded.
by Voip support 18 Nov '19

18 Nov '19

I am using kamailio 5.3. Ovidiu Sas, i will try this tomorrow today is bit late. Maybe it could work. David Villasmil , i know i can use SQLOPS but i was thinking of easier method of doing this. By successful REGISTER i mean save only REGISTER attempt that authenticated correctly at SIP server and in 200 OK a binding was returned. Same if the binding is later removed i would like remove it from database. I need this because when i run REGISTER over kamailio to destination SIP server i see all register from same IP. I need to be able to check what was the real source IP of client REGISTER. Regards, Tomasz pt., 15 lis 2019 o 00:16 Mike from ApoloCore <mike(a)apolocore.com> napisał(a): > I am using kamailio 5.3. > > Ovidiu Sas, i will try this tomorrow today is bit late. Maybe it could > work. > > David Villasmil , i know i can use SQLOPS but i was thinking of easier > method of doing this. > By successful REGISTER i mean save only REGISTER attempt that > authenticated correctly at SIP server and in 200 OK a binding was returned. > Same if the binding is later removed i would like remove it from database. > I need this because when i run REGISTER over kamailio to destination SIP > server i see all register from same IP. > I need to be able to check what was the real source IP of client REGISTER. > > Regards, > Tomasz > > > czw., 14 lis 2019 o 23:33 Ovidiu Sas <osas(a)voipembedded.com> napisał(a): > >> Try to use save() when you process the 200 ok reply for REGISTER in >> the reply route. >> >> Regards, >> Ovidiu Sas >> >> On Thu, Nov 14, 2019 at 3:40 PM Voip support <voipexpert0(a)gmail.com> >> wrote: >> > >> > I setup kamailio to forward REGISTER packets using dispatcher to two >> servers main and backup. >> > >> > The routing logic is quite simple i use the dispatcher probing >> mechanism to detect if my main dispatcher set id (one server) is active or >> inactive once it detect a failure all packets are sent to secondary SIP >> server to achieve High Availability. >> > >> > I am looking for module or a way to save into kamailio database client >> information such as IP address, register username (from FROM header), >> digest username (authorization username), call-id for successful >> registration. >> > >> > I tried to use registrar module but once i used function >> save("location") the client was registered on kamailio and packets was not >> forwarded to main/secondary SIP server (like dispatcher stopped working, >> the destination did not override). >> > >> > REGISTER is sent to SIP URI of kamailio. >> > >> > Please let me know if you have any idea how to make this data update in >> database (insert an entry on fresh successfull REGISTER, delete on expired >> or removed binding). >> > >> > Best regards, >> > Tomasz >> > >> > _______________________________________________ >> > Kamailio (SER) - Users Mailing List >> > sr-users(a)lists.kamailio.org >> > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> >> >> -- >> VoIP Embedded, Inc. >> http://www.voipembedded.com >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users(a)lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >

3 2

Saving REGISTER status in kamailio database while REGSITER packets are forwarded.
by Voip support 15 Nov '19

15 Nov '19

I setup kamailio to forward REGISTER packets using dispatcher to two servers main and backup. The routing logic is quite simple i use the dispatcher probing mechanism to detect if my main dispatcher set id (one server) is active or inactive once it detect a failure all packets are sent to secondary SIP server to achieve High Availability. I am looking for module or a way to save into kamailio database client information such as IP address, register username (from FROM header), digest username (authorization username), call-id for successful registration. I tried to use registrar module but once i used function save("location") the client was registered on kamailio and packets was not forwarded to main/secondary SIP server (like dispatcher stopped working, the destination did not override). REGISTER is sent to SIP URI of kamailio. Please let me know if you have any idea how to make this data update in database (insert an entry on fresh successfull REGISTER, delete on expired or removed binding). Best regards, Tomasz

5 5

dispatcher, pvar_hash and multiple kamailio instances
by Barry Flanagan 15 Nov '19

15 Nov '19

Hi, I have multiple Kamailio instances using dispatcher using alg “7” - hash over the content of PVs string, both using the same dispatcher list. Is it reasonable to expect both Kamailio, given the same hash_pvar, to select the same host from the same set? Looking at the source I do not see any reason why not, but I am getting different results from the two Kamailio servers. The only difference between them is different minor version numbers - 4.3.4 and 4.3.7 I want to ensure that calls with the same hash_pvar value go to the same backend no matter which Kamailio processes the request. Thanks! -Barry Flanagan

3 2

Re: [SR-Users] Question on Database Version Table
by JR Richardson 15 Nov '19

15 Nov '19

Yep version_table Set the name of the table holding the table version. Useful if the proxy is sharing a database within a project and during upgrades. Default value is “version”. Example of usage: version_table="version44" That's what I was looking for, perfect! Yes, I added accountcode and notes to my table for internal use, was just making notes on columns I need on the new version upgrade. Thanks. JR On Tue, Jun 18, 2019 at 2:17 PM Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > > Hello, > > like Alex said, you can have two version tables, one for each versions > of kamailio you play with and use in config the core parameter to set > the name of that table. > > I am actually writing to say that address table never had accountcode > and notes columns. Likely they were added in your deployment for other > purposes or either the default column names were changed. The default > column names are listed at: > > - > https://kamailio.org/docs/db-tables/kamailio-db-5.2.x.html#gen-db-address > > These were the same for rather long time, iirc the recent version number > changed mainly due to with modifications done to the size, not due to > new columns. > > Cheers, > Daniel > > On 18.06.19 20:50, JR Richardson wrote: > > Hey All, > > > > I'm doing some upgrades and noticed database table version numbers > > have increased. > > > > database table adjustments: > > | address | production ver 3 | upgrade 5.2 ver 6 | need to add > > accountcode and notes column > > | dispatcher | production ver 3 | upgrade 5.2 ver 4 | need to add attr column > > | trusted | production ver 5 | upgrade 5.2 ver 6 | need to add > > ruri_pattern and priority column > > > > I run mysql master/slave so I'm concerned about the master database > > not having the correct version numbers or can I just update the > > version number in the table when I do the upgrade of the kamailio > > versions? > > > > Thanks. > > > > JR

4 6

Processes blocking - troubleshooting slow actions
by Julien Chavanton 15 Nov '19

15 Nov '19

I am troubleshooting intermittent latency on a server, several processes will report slow action at the same time every now and then, something odd it that some of the actions are most likely not using any mutexes, I can not imagine how they can end up blocking unless the process is actually suspended by the OS, however the CPU usage and load average is quite low on this server. Is there any reason why assigning to a PKG var could be delayed 3 seconds. 2019-11-13T21:58:04.018631614Z 23(42) WARNING: <core> [core/action.c:1586]: run_actions(): alert - action [corefunc (16)] cfg [/etc/kamailio/kamailio.cfg:115] took too long [2579486 us] $var(reply) = 0; // I wonder how this can endup blocking ? 2019-11-13T21:58:04.018748710Z 42(61) WARNING: <core> [core/action.c:1586]: run_actions(): alert - action [corefunc (16)] cfg [/etc/kamailio/kamailio.cfg:221] took too long [2222010 us] } // Samething here, simply exiting a scope 2019-11-13T21:58:04.015675095Z 11(30) WARNING: <core> [core/action.c:1586]: run_actions(): alert - action [t_relay_cancel (24)] cfg [/etc/kamailio/kamailio.cfg:86] took too long [2588818 us] } if (!t_relay_cancel()) { // Indeed a t_relay_cancel 2019-11-13T21:58:04.018647988Z 26(45) WARNING: <core> [core/action.c:1586]: run_actions(): alert - action [t_relay (24)] cfg [/etc/kamailio/kamailio.cfg:289] took too long [2739366 us] t_relay(); // Indeed a t_relay 2019-11-13T21:58:04.018651243Z 15(34) WARNING: <core> [core/action.c:1586]: run_actions(): alert - action [sl_send_reply (26)] cfg [/etc/kamailio/kamailio.cfg:112] took too long [2587812 us] sl_send_reply("100", "Trying"); // Indeed a sl_send_reply Not sure which mailing list is best for this question Regards Julien

2 4

kamailio 5.2.4 unable to load db_mysql.so
by Rupesh Kumar 15 Nov '19

15 Nov '19

I am trying to compile and load kamailio 5.2.4 on centos 7.5. I am getting below errors whenever I start kamailio . I have the latest openssl and openssl-devel packages installed on my centos . Not sure what other dependencies . Not starting Kamailio SIP Server: invalid configuration fil[FAILED] [FAILED]0(23235) ERROR: <core> [core/sr_module.c:498]: load_module(): could not open module </usr/local/lib64/kamailio/modules/db_mysql.so>: /usr/local/lib64/kamailio/modules/db_mysql.so: undefined symbol: SSL_get_error 0(23235) CRITICAL: <core> [core/cfg.y:3508]: yyerror_at(): parse error in config file //usr/local/etc/kamailio/kamailio.cfg, line 229, column 12-24: failed to load module ERROR: bad config file (1 errors) 0(23235) INFO: <core[FAILED]sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized [FAILED]

1 1

Next online devel meeting - irc/video conference - Thursday, Nov 14, 15:00UTC
by Daniel-Constantin Mierla 14 Nov '19

14 Nov '19

Hello, on Thursday and Friday this week over a dozen of Kamailio developers will be in Dusseldorf for the annual coding meeting, so we thought of doing an online meeting via some video conferencing bridge, with IRC as a backup chatting channel, to allow also other community members to join remotely and discuss the development plans for short/medium term. It is more or less what we used to do during the past IRC devel meetings, but to make the discussions go faster and shorter the meeting, so people in Dusseldorf spend more time coding, this time we will try video conferencing. Very likely we will use Jitsi Meet, so anyone that wants to join should be prepared with a WebRTC capable web browser (all major ones support it) -- link and details how to join will be provided short before the meeting starts, right now the plan is to do it on Thursday, Nov 14, at 15:00UTC/London (16:00Berlin-Paris, 10:00New York). If someone wants to address questions but doesn't have the possibility to join the video conf bridge, then join the IRC #kamailio channel on freenode.net and post your question there during the meeting. The IRC channel will be monitored and used for text chatting whenever needed. 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

2 3

Kamailio v5.3.1 Released
by Daniel-Constantin Mierla 14 Nov '19

14 Nov '19

Hello, Kamailio SIP Server v5.3.1 stable release is out. This is a maintenance release of the latest stable branch, 5.3, that includes fixes since the release of v5.3.1. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.3.x. Deployments running previous v5.3.x versions are strongly recommended to be upgraded to v5.3.1. For more details about version 5.3.1 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2019/11/kamailio-v5-3-1-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! 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

1 0

Releasing Kamailio v5.3.1
by Daniel-Constantin Mierla 14 Nov '19

14 Nov '19

Hello, the plan right now is to release v5.3.1 during the next Kamailio Developers Meeting, if we succeed to coordinate on site, therefore expect it to happen next week during Nov 14-15. Besides just doing this specific release, to goal is to try to build a more automatic process for releases, document how is done, etc... updates about the results will follow after the event. Anyhow, for now the usual the pre-release notice: if anyone is aware of some issue that is not yet reported, do it asap in order to give it a chance to be analyzed and fixed. 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

1 1

event_route[core:msg-received] native vs kemi/lua
by GG GG 14 Nov '19

14 Nov '19

Hi! I just made a simple test performance between native script and kemi/lua with the event_route[core:msg-received] callback. # lua script children=1 cpu.process=100% function ksr_core_received() exit; end # native script children=1 cpu.process=70% event_route[core:msg-received] { exit; } # tested with udp 5060, 1000 Mbits/sec iperf -c 127.0.0.1 -p 5060 -u -b1000m I don't know if it's normal and what we should expect from kemi, but I would like to share this with you. Maybe someone could confirm the performance difference. Have a good day !

2 8

kemi framework design
by GG GG 12 Nov '19

12 Nov '19

Hi, looking at the documentation and the code, I couldn't answer myself. - does kamailio use one lua interpreter per fork ? - about the safety, I just made a bad lua module in c, to force a segmentation fault in the module. I would expect kamailio to protect itself from the crash, but kamailio restarted.

2 1

Change SDP Session Name
by Alvaro Araos 11 Nov '19

11 Nov '19

Hi everyone, Exists some way to change the SDP Session Name value (s=)??? Saludos, Alvaro Araos

3 2

Change SDP Session Name
by Alvaro Araos 11 Nov '19

11 Nov '19

Hi everyone, Exists some way to change the SDP Session Name value (s=)??? Saludos, Alvaro Araos

1 0

Change SDP Session Name
by Alvaro Araos 11 Nov '19

11 Nov '19

Hi everyone, Exists some way to change the SDP Session Name value (s=)??? Saludos, Alvaro Araos

1 0

Kamailio in cloud environments
by Alejandro Recarey 11 Nov '19

11 Nov '19

Hi, I am trying to get kamailio working cloud provider, autoscaling them behind a TCP load balancer. Is there a way for kamailio to discover its Public IP at startup? Or do people generally program their own startup scripts that modify the kamailio config file before starting it? Is there any gotchas you have hit while configuring this setup? Thank you! Alex

8 13

uac_replace_from(uri) alters display name
by Sergiu Pojoga 10 Nov '19

10 Nov '19

Hi ppl, Need to replace origination carrier's domain with my own, so I do: branch_route[MANAGE_PSTN_IN_BRANCH] { uac_replace_from("sip:$fU@mydomain.net"); } The dialog-forming INVITE is relayed properly with ONLY the domain part being replaced. However, subsequent messages such as ACK and BYE have the From DISPLAY NAME (_uac_fdp) stripped off. Not that it affects real functionality in any known way (CLID is still good for callee), but curious if this is intentional or not, as *uac_replace_from(uri)* clearly states that: "*Replace in FROM header the URI part without altering the display name*" # kamcmd dlg.list variables: { { _uac_fdpnew: } { _uac_fdp: "JOHN DOE" } { _uac_funew: sip:18661112222@mydomain.net } { _uac_fu: sip:18661112222@carrierdomain.net } Thanks!

2 4

OPTIONS request not authenticating
by Gustavo A. M. 10 Nov '19

10 Nov '19

Greetings, I'm trying to configure kamailio to get OPTIONS response, but it does not working properly. After a successful REGISTER, I send OPTIONS first time, get Proxy- Authenticate header to calculate Proxy-Authorization, and resend OPTIONS with Proxy-Authenticate header, but server follows answering with a 407 response, apparently ignoring Proxy-Authenticate. Note that same password is used in REGISTER request, and first time REGISTER response with a 401, not a 407 response. I think messages sent from client are right, and problem is on the server configuration. Follow messages and kamailio.cfg. Thanks in advance, Gustavo Adolfo Moennich OPTIONS sip:6000@kamailio.memorea.com SIP/2.0 Call-ID: A6DB9DCAA18049E784034FB6F5F95680 Contact: <sip:6000@kamailio.memorea.com> Content-Length: 0 CSeq: 0 OPTIONS Expires: 20 From: "Bob" <sip:6000@kamailio.memorea.com>;tag=0B3D2156FF9E46DAAD7FC43 B62C9E2FC Max-Forwards: 70 To: "Bob" <sip:6000@kamailio.memorea.com> User-Agent: Memorea 1.0 (SIP 2.0) Build 1.1.1.0, Windows x64 Build 6.1.7601 Via: SIP/2.0/UDP 192.168.122.21:5060;rport=58905 SIP/2.0 407 Proxy Authentication Required Call-ID: A6DB9DCAA18049E784034FB6F5F95680 CSeq: 0 OPTIONS From: "Bob" <sip:6000@kamailio.memorea.com>;tag=0B3D2156FF9E46DAAD7FC43 B62C9E2FC To: "Bob" <sip:6000@kamailio.memorea.com>;tag=9dd61ff61e802d8e2bef5f146 21ef3c2.467d Via: SIP/2.0/UDP 192.168.122.21:5060;rport=58905;received=192.168.122.12 Proxy-Authenticate: Digest realm="kamailio.memorea.com", nonce="Xcc1wV3HNJXdfUDchNlEUkT6kHCtqBAn" Server: kamailio (5.2.1 (x86_64/linux)) Content-Length: 0 OPTIONS sip:6000@kamailio.memorea.com SIP/2.0 Call-ID: A6DB9DCAA18049E784034FB6F5F95680 Contact: <sip:6000@kamailio.memorea.com> Content-Length: 0 CSeq: 1 OPTIONS Expires: 20 From: "Bob" <sip:6000@kamailio.memorea.com>;tag=0B3D2156FF9E46DAAD7FC43 B62C9E2FC Max-Forwards: 70 Proxy-Authorization: Digest uri=sip:6000@kamailio.memorea.com,nonce="Xc c1wV3HNJXdfUDchNlEUkT6kHCtqBAn",realm="kamailio.memorea.com",response=" 5bcd184eea53284f3ec548c2412ce36a",username=6000 To: "Bob" <sip:6000@kamailio.memorea.com> User-Agent: Memorea 1.0 (SIP 2.0) Build 1.1.1.0, Windows x64 Build 6.1.7601 Via: SIP/2.0/UDP 192.168.122.21:5060;rport=58905 SIP/2.0 407 Proxy Authentication Required Call-ID: A6DB9DCAA18049E784034FB6F5F95680 CSeq: 1 OPTIONS From: "Bob" <sip:6000@kamailio.memorea.com>;tag=0B3D2156FF9E46DAAD7FC43 B62C9E2FC To: "Bob" <sip:6000@kamailio.memorea.com>;tag=9dd61ff61e802d8e2bef5f146 21ef3c2.32de Via: SIP/2.0/UDP 192.168.122.21:5060;received=192.168.122.12;rport=58905 Proxy-Authenticate: Digest realm="kamailio.memorea.com", nonce="Xcc1xl3HNJqRobJFuZggPSGvNdtp+GnP" Server: kamailio (5.2.1 (x86_64/linux)) Content-Length: 0 kamailio.cfg content's: #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB # # Kamailio (OpenSER) SIP Server v5.2 - default configuration script # - web: https://www.kamailio.org # - git: https://github.com/kamailio/kamailio # # Direct your questions about this file to: <sr-users(a)lists.kamailio.or g> # # Refer to the Core CookBook at https://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Note: the comments can be: # - lines starting with #, but not the pre-processor directives, # which start with #!, like #!define, #!ifdef, #!endif, #!else, #!trydef, # #!subst, #!substdef, ... # - lines starting with // # - blocks enclosed in between /* */ # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To block 401 and 407 authentication replies execute: # - define WITH_BLOCK401407 # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:temp@database.memorea.com/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes */ children=8 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases * based on reverse DNS on IPs (default on) */ # auto_aliases=no /* add local domain aliases */ # alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to * bind on a specific interface/port/proto (default bind on all available) */ # listen=udp:10.0.0.10:5060 #!ifdef WITH_TLS enable_tls=yes #!endif /* life time of TCP connection when there is no traffic * - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ #!ifdef WITH_PSTN /* PSTN GW Routing * * - pstn.gw_ip: valid IP or hostname as string value, example: * pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" * * - by default is empty to avoid misrouting */ pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL /* VoiceMail Routing on offline, busy or no answer * * - by default Voicemail server IP is empty to avoid misrouting */ voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ # modparam("registrar", "max_contacts", 10) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- /* ip ban htable with autoexpire after 5 minutes */ modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=$si) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if (uri==$ru) { if ((method==OPTIONS) && (! uri=~"sip:.*[@]+.*")) { options_reply(); } } if(!sanity_check("17895", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == $ru) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) { sl_reply_error(); } exit; } # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==$fu) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } if (is_method("OPTIONS") || from_uri==$fu) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("OPTIONS|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=$fu && uri!=$ru) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=$ru) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==$ru) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=$fu) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + if (starts_with("$rU", "00")) { strip(2); prefix("+"); } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif }

2 1

dispatch REGISTER over TLS
by sthustfo 10 Nov '19

10 Nov '19

I have a basic setup where kamailio receives SIP over websocket (no WSS) and forwards to SIP server over TLS. I have enabled TLS in kamailio.cfg and added dispatcher node as sips:SIP_SERVER:5061 and transport=tls. +----+-------+------------------------+-------+----------+---------------+----------------+ | id | setid | destination | flags | priority | attrs | description | +----+-------+------------------------+-------+----------+---------------+----------------+ | 4 | 1 | sips:10.0.0.100:5061 | 0 | 0 | transport=tls | SIP SERVER | +----+-------+------------------------+-------+----------+---------------+----------------+ Now when REGISTER is received over websocket, kamailio is responding with error code 500 and phrase "500 I'm terribly sorry, server error occurred (7/SL)". And on the console I see the following error messages. 12(33858) DEBUG: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} <core> [core/md5utils.c:67]: MD5StringArray(): MD5 calculated: f1ecf7bcb659b07fe81e332e100044e5 12(33858) ERROR: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} tm [ut.h:315]: uri2dst2(): no corresponding socket found for "10.0.0.100" af 2 (tls: 10.0.0.100:5061) 12(33858) ERROR: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} tm [t_fwd.c:467]: prepare_new_uac(): can't fwd to af 2, proto 3 (no corresponding listening socket) 12(33858) ERROR: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} tm [t_fwd.c:1735]: t_forward_nonack(): failure to add branches 12(33858) DEBUG: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} tm [t_funcs.c:334]: t_relay_to(): t_forward_nonack returned error -7 (-7) 12(33858) DEBUG: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} tm [t_funcs.c:352]: t_relay_to(): -7 error reply generation delayed 12(33858) exec: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} *** cfgtrace:request_route=[RELAY] c=[/etc/kamailio/kamailio.cfg] l=587 a=24 n=sl_reply_error 12(33858) DEBUG: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} <core> [core/msg_translator.c:162]: check_via_address(): (10.0.0.14, hsvmphm3ps12.invalid, 0) 12(33858) DEBUG: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} websocket [ws_conn.c:452]: wsconn_get(): wsconn_get for id [3] *tls.cfg contents* [client:default] method = TLSv1 verify_certificate = yes require_certificate = yes private_key = /home/test/kamailio/internal.key certificate = /home/test/kamailio/internal.crt ca_list = /home/test/kamailio/ca_list.pem Any reason why this error is seen? Any inputs appreciated. Thanks.

4 20

Re: [SR-Users] _w_t_relay_to(): t_forward_noack failed when forwarding call
by Konstantinos Merentitis 08 Nov '19

08 Nov '19

I added $du=$null; before route(RELAY) as you suggested but I get the same results: Nov 8 12:40:22 spock /usr/sbin/kamailio[21826]: ERROR: <script>: transaction canceled Nov 8 12:40:22 spock /usr/sbin/kamailio[21826]: ERROR: <script>: #012 RDNIS = sip:297@192.168.1.134 Nov 8 12:40:22 spock /usr/sbin/kamailio[21826]: ERROR: <script>: =============forwarding to voicemail1, RDNIS: sip:297@192.168.1.134 =========== Nov 8 12:40:22 spock /usr/sbin/kamailio[21826]: ERROR: tm [tm.c:1679]: _w_t_relay_to(): t_forward_noack failed Nov 8 12:40:22 spock /usr/sbin/kamailio[21826]: ERROR: sl [sl_funcs.c:392]: sl_reply_error(): stateless error reply used: transaction canceled (487/SL) Regards, Konstantinos > On Fri Nov 8 10:24:06 CET 2019 Daniel Tryba wrote: > >> On Thu, Nov 07, 2019 at 04:53:43PM +0200, Konstantinos Merentitis wrote: >> >> I tried to modify the failure route in order to also forward not answered (failed?) calls to asterisk mailbox but i get the following error and the call is never forwarded: >> > You appear to do the same thing I do, except setting $du to null. If the user is REGISTERed $du will be set to their endpoint, changing $ru will have no effect on routing. So add $du=$null; in the TOVOICEMAIL route before route(RELAY) and see what happens. >

3 4

_w_t_relay_to(): t_forward_noack failed when forwarding call
by Konstantinos Merentitis 08 Nov '19

08 Nov '19

I would like to forward all incoming calls that are not answered from kamailio, to my asterisk voicemail server. - I have not setup any Realtime Integration. I am using the "Redirected Dialed Number Information Service" header and calling "554(a)192.168.1.134" which is my voicemail server. Forwarding succeeds when the subscriber is not registered to kamailio: Nov 7 16:19:45 spock /usr/sbin/kamailio[17606]: ERROR: <script>: =============User sip:6297@192.168.1.131 not found here, going to voicemail============= Nov 7 16:19:45 spock /usr/sbin/kamailio[17606]: ERROR: <script>: #012 RDNIS = sip:297@192.168.1.134 Nov 7 16:19:45 spock /usr/sbin/kamailio[17606]: ERROR: <script>: =============forwarding to voicemail1, RDNIS: sip:297@192.168.1.134 =========== I tried to modify the failure route in order to also forward not answered (failed?) calls to asterisk mailbox but i get the following error and the call is never forwarded: Nov 7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: <script>: transaction canceled Nov 7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: <script>: #012 RDNIS = sip:297@192.168.1.134 Nov 7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: <script>: =============forwarding to voicemail1, RDNIS: sip:297@192.168.1.134 =========== Nov 7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: tm [tm.c:1679]: _w_t_relay_to(): t_forward_noack failed Nov 7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: sl [sl_funcs.c:392]: sl_reply_error(): stateless error reply used: transaction canceled (487/SL) failure and voicemail routes are slightly modified from the default config: route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { xlog("=============User $ru not found here, going to voicemail=============\n"); $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { $var(RDNIS)=$di; xlog("\n RDNIS = $var(RDNIS) \n"); if($avp(oexten)==$null) return; # $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip + ";transport=udp") # + ":" + $sel(cfg_get.voicemail.srv_port); xlog("=============forwarding to voicemail1, RDNIS: $di =========== "); $ru = "sip:554@192.168.1.134:5060"; } else { if($rU==$null) return; # $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip + ";transport=udp") # + ":" + $sel(cfg_get.voicemail.srv_port); xlog("=============forwarding to voicemail2, RDNIS: $di =========== "); $ru = "sip:554@192.168.1.134:5060"; } route(RELAY); exit; #!endif return; } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { #modified here xlog("transaction canceled\n"); route(TOVOICEMAIL); #exit; #/modified } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; xlog("============= Route failure, forwarding to voicemail ============="); route(TOVOICEMAIL); exit; } #!endif }

2 1

How to read some data inside the INVITE ..
by Laura 07 Nov '19

07 Nov '19

Dears.. how is possibile to read that data 9052562462354464 inside the first INVITE message === Example === U 2019/11/07 14:46:59.320936 192.168.1.1:5060 -> 192.168.1.242:5060 INVITE sip:9052562462354464@192.168.1.242 SIP/2.0. Record-Route: <sip:192.168.1.1;lr;ftag=105854496;did=24e.e751;vst=AAAAAAAAAwQAAAAAAAAAAAAAAAAADQIAAwEDAAAHDwAGBAg1MDYw;nat=yes>. Max-Forwards: 66. Via: SIP/2.0/UDP 51.254.158.40:5060;branch=z9hG4bK501c.87a8e402c8f1cf93192ddd8e6744da57.1. From: "38617774738" <sip:38617774738@10.1.1.1>;tag=105854496. To: <sip:562462354464@192.168.1.242>. Call-ID: 1573138018-1602381967(a)10.1.1.182. CSeq: 20215490 INVITE ..... ETC ETC... Thanks to all

4 4

Re-invite after RTPengine pacemaker switching?
by Жан Базаров 07 Nov '19

07 Nov '19

I need to send re-invite after pacemaker fails over on new rtpengine server. Because new rtpengine dont participate in DTLS handshake and i hear nothing, but silence. I think, may me its would be work. Do you have any idea on this issue?

3 5

multihomed kamailio and sip-reinvites SRTP to RTP
by Karsten Horsmann 07 Nov '19

07 Nov '19

Hello Mailinglist, i try to figure out, how to solve RE-INVITES and SRTP: i have an kamailio route names route[MEDIAPROXY] that i used to controll rtpengine in an private and public ip setup. This works fine for me now. It use the permission module and the adress table to figure out $var(mediagroup) = allow_address_group($Ri, 0); received from internal or external ip. The idea is from SBC-OS chamless stolen, so thanks for that. So fare so good. I adapted SRTP also to this. But on RE-INVITES it stops handling SRTP correctly to the encryption upstream. And i am sure, its my fault. But i have no clue how to detect that correctly. Any hints where i can read more about an correct handling on this? Here my main-config file / kamailio 5.3.x and rtpengine. https://gist.github.com/khorsmann/02bd98a628a190caee68fc3d876a3448 -- Mit freundlichen Grüßen *Karsten Horsmann*

2 2

Re: [SR-Users] Group module
by Daniel Tryba 07 Nov '19

07 Nov '19

On Wed, Nov 06, 2019 at 07:15:42PM +0100, Igor Olhovskiy wrote: > > Hm... Maybe there is other module to achieve such functions? > Best if it would be with cache :) > But if no - regex also fine. But if it works it works! I don't think there is a specific module to do this but it is easy to implement yourself. What I don't like personally about your solution is the use of the permission tables, you need atleast 1 per user (unless users have the same ip). Personally I do this by direct database queries in usr_preferences on INVITEs and REGISTERs. You could use any database (depending on how you store the ACL) and cache it in a htable. My solution is mysql specific (inet_aton) and forces users to have atleast 1 CIDR notated subnet: if(!avp_db_query("select value from usr_preferences where username='$au' and attribute='acl' and inet_aton(substring_index(value,'/',1))&(1 << 32) - 1 & ~((1 << (32 - substring_index(value,'/',-1))) - 1)=inet_aton('$si')&(1 << 32) - 1 & ~((1 << (32 - substring_index(value,'/',-1))) - 1)")) { t_reply("404","Go away"); exit; }

3 2

MongoDB findone(). Is projection supported?
by Lars Olsson 06 Nov '19

06 Nov '19

Hi, I am using NDB_MONGODB to ask some questions to a database. My database structure is pretty heavy, so I like to return as little data as required to Kamailio. I have tried the following MongoDB syntax for projection to include only given (my_field) column in the response. mongodb_find_one("srv_name", "database", "collection", "{ \"user_id\" : \"$var(user_id)\" }, { \"my_field\" : 1}", "mgr1") I always get full DB entry back. Is this supported? The same string works OK in the vanilla mongo client. How can I limit the result? Thanks Lars

2 3

Group module
by Igor Olhovskiy 06 Nov '19

06 Nov '19

Hi! I'm trying to solve quite simple task and want to solve it as simple as possible. Idea is to have restriction to allow register certain username/domain only from certain subnets. So, I decided to use group and permissions modules. But stuck on group one. Code is fairly simple modparam("group", "db_url", DBURL) modparam("group", "use_domain", 1) ... if (get_user_group("From", "$avp(user_groups)")) { xlog("$fu from $si:$sp is matched group $avp(user_groups)\n"); if (allow_source_address("$avp(user_groups)")) { ...allow furter } } else { xlog("$fu from $si:$sp is not matched any groups\n"); } grp table is looks like kamctl db exec 'select * from grp' +----+-------------+-------------------+-----+---------------------+ | id | username | domain | grp | last_modified | +----+-------------+-------------------+-----+---------------------+ | 1 | 61388140395 | sip.localhost.net | 5 | 2000-01-01 00:00:01 | +----+-------------+-------------------+-----+---------------------+ But actual line of debug in logs saying this kamailio_1 | 10(18) exec: {1 1 REGISTER 4oQTNaaVlEWQC2VMK6YAKg..} *** cfgtrace:request_route=[PERMISSIONS] c=[/etc/kamailio/kamailio.cfg] l=720 a=26 n=get_user_group kamailio_1 | 10(18) DEBUG: {1 1 REGISTER 4oQTNaaVlEWQC2VMK6YAKg..} group [re_group.c:188]: get_user_group(): getting groups for <sip:61388140395@sip.localhost.net> kamailio_1 | 10(18) exec: {1 1 REGISTER 4oQTNaaVlEWQC2VMK6YAKg..} *** cfgtrace:request_route=[PERMISSIONS] c=[/etc/kamailio/kamailio.cfg] l=727 a=25 n=xlog kamailio_1 | 10(18) ERROR: {1 1 REGISTER 4oQTNaaVlEWQC2VMK6YAKg..} <script>: sip:61388140395@sip.localhost.net;transport=UDP from 172.21.0.1:56787 is not matched any groups What am I missing here? Or module is too old? Thanks!

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