- sr-users - kamailio.org

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

03 Feb '20

Hey guys - I'm using the following settings on my Kamailio config: modparam("tls", "low_mem_threshold2", 1024) modparam("tls", "tls_force_run", 1) modparam("tls", "tls_disable_compression", 1) modparam("tls", "ssl_max_send_fragment", 4096) modparam("tls", "renegotiation", 0) modparam("tls", "low_mem_threshold1", 2048) modparam("tls", "require_certificate", 1) When a client attempts to connect to Kamailio, the server hello it receives does *not* contain the client certificate request. And I've noticed that kamailio starts with: tls [tls_domain.c:694]: set_verification(): TLSs<default>: No client certificate required and no checks performed tls [tls_domain.c:320]: fill_missing(): TLSs<default>: require_certificate=0 Complete version output: kamailio 5.1.1 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. Any hints?

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

03 Feb '20

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

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sqlops multiple databases
by Ali Taher 03 Feb '20

03 Feb '20

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

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

31 Jan '20

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

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

31 Jan '20

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

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

31 Jan '20

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

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

31 Jan '20

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

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kamailio
by Genildo Nascimento 30 Jan '20

30 Jan '20

Someone managed installed kamailio with IPV6

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

30 Jan '20

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

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

29 Jan '20

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

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

1 0

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

2 1

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" )

4 6

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

5 8

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 >

2 1

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.

1 0

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

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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/

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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 >> >

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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

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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

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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

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