sr-users July 2020

sr-users@lists.kamailio.org

58 participants
89 discussions

Change FROM to hide topology
by Andre Pires dos Santos 05 Sep '20

05 Sep '20

Hi guys, We have SIP Server > Kamailio > webrtc client. The only call flow we have is SIP server calling kamailio tha sends the call to webrtc client. We set up TOPOH to hide the sip server info but once the sip server initiates the call we still have the FROM details identifying SIP Server IP address. We couldn't find a way to mask this ip using TOPOH. What would be the best way to hide the sip server IP address in this case? Thanks a lot! Andre

3 2

Route to PSTN no works
by sip user 07 Aug '20

07 Aug '20

Hi, I have kamailio connect to Teams, and works form Asterisk -> Teams calls. For Teams -> Asterisk calls I'd worked using extension and register Asterisk with that extension. But I'd like to use direct routing with IP. In kamailio.cfg I activate define WITH_PSTN. I configured the IP and PORT for my PSTN. I'm using the default route[PSTN]: route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined xlog("L_INFO","PSTN ACTIVADO"); if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")){ xlog("L_INFO", "Error en el formato numerico!!"); return; } # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + #if (starts_with("$rU", "00")) { # strip(2); # prefix("+"); #} if (strempty($sel(cfg_get.pstn.gw_port))) { #$ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); xlog("L_INFO","SELECCION CON PUERTO"); $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } else { xlog("L_INFO","SELECCION CON PUERTO"); $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } And in my request_route: remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { if($src_ip != "IP ASTERISK"){ xlog("L_INFO", "***********ROUTE PSTN***********"); route(PSTN); } else { xlog("L_INFO","LLamada desde $si con puerto $sp"); record_route_preset("FQND:5061;transport=tls", "IP KAMAILIO:5060"); add_rr_param(";r2=on"); route(DISPATCH); route(RELAY); } } But never see that the call go to PSTN route.. I'd made any wrong?? Thanks

3 2

ACC events missing to-tag when doing t_reply
by Julien Chavanton 04 Aug '20

04 Aug '20

Hi, considering this example : modparam("acc_json", "acc_missed_flag", 6) failure_route[test] { setflag(6); // MISSED_ACC, the failure will be recorded by ACC t_flush_flags(); // This will set the flags in the newly created transaction t_reply("500", "Service Unavailable"); } This does generate an ACC event, however the to-tag is not populated, my guess this is because the TM callback is done before the to-tag is generated. I wonder if there is a way to populate the to-tag immediately ? I will dig it further, but maybe there is a trick I am missing

2 2

Kamailio Dockerization
by Edward Romanenco 04 Aug '20

04 Aug '20

Hi, I am working on a project involving Kamailio dockerezation, which is meant to run alongside Freeswitch and RTPEngine containers, on the basis of a Docker-Compose file which is launched on top of a CentOS 7.7 host system. I was able to create and run the containers successfully, they are starting and listening to the correct ports, but for some unexplained reason - the incoming SIP traffic is not getting picked up by Kamailio. I can easily trace the traffic from the host, but when SSHing the container and running a test from within, no traffic goes by. I've used netcat to generate plain UDP traffic to the container, and it was logged into the Kamailio log files, but real-life traffic doesn't seem to work. I've tried moving to host mode (from bridge), but it didn't make any difference. All required firewall rules were opened obviously, I've also tried shutting the firewall off completely but it didn't help. Does anyone experienced anything similar while running Kamailio in Dockers, and could provide me a go-through on what steps did he take to fix it? EXCERPT FROM MY DOCKERFILE # Getting Kamailio source code from GIT RUN mkdir -p /usr/local/src/kamailio-5.3 WORKDIR /usr/local/src/kamailio-5.3 RUN git clone --depth 1 --no-single-branch https://github.com/kamailio/kamailio WORKDIR /usr/local/src/kamailio-5.3/kamailio RUN git checkout -b 5.3.2 # Compile the source code and install Kamailio RUN make include_modules="phonenum db_mysql xmlrpc http_async_client jansson auth_db nathelper websocket tls outbound topoh http_client" cfg && \ make all && make install # Default setting is to run Kamailio as user “kamailio” and group “kamailio” RUN adduser --quiet --system --group --disabled-password \ --shell /bin/false --gecos "Kamailio" \ --home /var/run/kamailio kamailio # To use init.d script for starting/stopping the Kamailio server COPY Init/kamailio /etc/init.d/ RUN chmod 755 /etc/init.d/kamailio COPY Default/kamailio /etc/default/ COPY kamailio.service /etc/systemd/system/ RUN mkdir -p /var/run/kamailio RUN chown kamailio:kamailio /var/run/kamailio COMPOSE (BRIDGE NETWORK VERSION) kamailioegress: build: kamailio_egress image: kamailioegress:latest container_name: kamailioegress restart: always environment: - DATABASE=kamailioe - SIP_DOMAIN=XXX - DBHOST=kamailiodb - DBROOTUSER=root - DBROOTPASS=XXX - PUBLIC_IPV4=XXX depends_on: - Kamailio-Base - kmdb - freeswitch - rtpengine expose: - "5060/udp" - "5060/tcp" ports: - "XXX:5060:5060/udp" networks: private-net: ipv4_address: "172.18.0.30" deploy: mode: replicated replicas: 1 restart_policy: condition: always delay: 5s max_attempts: 3 window: 120s networks: private-net: driver: bridge ipam: config: - subnet: 172.18.0.0/16 driver_opts: com.docker.network.bridge.name: wrtcpriv public-net: external: name: host COMPOSE (HOST MODE VERSION) kamailioegress: build: kamailio_egress image: kamailioegress:latest container_name: kamailioegress network_mode: host restart: always environment: - DATABASE=kamailioe - SIP_DOMAIN=XXX - DBHOST=172.18.0.10 - DBROOTUSER=root - DBROOTPASS=XXX - PUBLIC_IPV4=XXX - EGPORT=5060 - LINTE=ens224 - LINTI=ens192 - RTPENGINE=localhost depends_on: - Kamailio-Base - kmdb - freeswitch - rtpengine expose: - "5060/udp" ports: - "213.8.76.13:5060:5060/udp" deploy: mode: replicated replicas: 1 restart_policy: condition: always delay: 5s max_attempts: 3 window: 120s CONFIG FILE /* uncomment and configure the following line if you want Kamailio to * bind on a specific interface/port/proto (default bind on all available) */ listen=udp:0.0.0.0:LPORT advertise PUBLIC_IP:LPORT KAMAILIO-LOCALE #!define DBURL "mysql://root:XXX@DBHOST/kamailioe" #!substdef "!MY_DBURL!mysql://root:XXX@DBHOST/kamailioe!g" #!substdef "!RTPENGINE!MY_RTPENGINE!g" #!substdef "!SIP_DOMAIN!MY_SIP_DOMAIN!g" #!substdef "!PUBLIC_IP!MY_PUBLIC_IP!g" #!substdef "!PRIVATE_IP!MY_PRIVATE_IP!g" #!substdef "!LPORT!MY_LPORT!g" #!substdef "!LINT!MY_LINT!g" #!substdef "!HOMER_IP!10.1.0.100!g" #!substdef "!API_URL!http://localhost:3000/v1/mock!g" #!substdef "/CCODES/972|380/" #!substdef "/NUM_TRANSLATE_OUT_RE/\\\\+?(CCODES)([0-9]+)/" #!substdef "/NUM_TRANSLATE_IN_RE/0([0-9]+)/" FIREWALL RULES -bash-4.2# firewall-cmd --list-all public (active) target: default icmp-block-inversion: no interfaces: ens192 ens224 sources: 192.168.1.39 services: dhcpv6-client http https sip ssh ports: 9323/tcp 9323/udp protocols: masquerade: no forward-ports: source-ports: icmp-blocks: rich rules: rule family="ipv4" destination address="XXX" port port="5060" protocol="udp" accept rule family="ipv4" destination address="XXX" port port="5060" protocol="tcp" accept NETSTAT OUTPUT Active Internet connections (only servers) Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name udp 0 0 XXX:5060 0.0.0.0:* 22479/kamailio Edward

4 7

[Help] Kamailio not receive bye requests
by Duy Phan 04 Aug '20

04 Aug '20

Hi guy! I use ec2 ubuntu on aws. But kamailio not receive bye requests. But when I install Kmailio in vultr is OK. can you help me?

4 3

search_append_body?
by Steve Wilkins 04 Aug '20

04 Aug '20

Hello, Does search_append_body require an "\r\n" in the attribute that is being added? If I don't use a "\r\n", it appears to look correct in the trace, however one provider says the attribute that was added was just a continuation of the prior attribute, which Wireshark does not show to be the case. Thank you,

2 1

Setting up MSFT Teams SBC <> SIP trunk but forwarding ACK and CANCEL messages
by Daniel van der Tang 02 Aug '20

02 Aug '20

Hello, I'm setting up a Kamailio MSFT Teams SBC that is connected to our SIP trunk provider. This server is directly connected to the Internet and not behind a NAT routing. It is successfully processing invites from both sides (MSFT Teams and SIP trunk). Unfortunately the ACKs and CANCEL messages are not relaying to Teams. In the sipdump i can't find these messages. Does anyone see what I am doing wrong? Below you can find my config #!KAMAILIO ####### Defined Values ######### #!define MULTIDOMAIN 0 # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 #!define FROM_TEAMS 11 #!define FROM_PBX 12 ######## Define Modules ########### #!define WITH_RTPENGINE #!define WITH_TLS #!define WITH_SIPDUMP #!define WITH_DISPATCH ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR debug=2 log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes */ children=2 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes alias=SBC_FQDN /* listen addresses */ listen=udp:PUBLIC_IP:5060 advertise SBC_FQDN:5060 #!ifdef WITH_TLS listen=tls:PUBLIC_IP:5061 advertise SBC_FQDN:5061 #!endif server_header= "Server: ABC SBC" user_agent_header= "User-Agent: ABC SBC" ###### TLS Enable ###### #!ifdef WITH_TLS enable_tls=yes #tcp_connect_timeout=1000 tcp_accept_no_cl=yes tcp_async = yes tcp_connection_lifetime=600 /* upper limit for TLS connections */ tls_max_connections=2048 #!endif ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ ####### Modules Section ######## /* set paths to location of modules */ loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" #!ifdef WITH_RTPENGINE loadmodule "rtpengine.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_SIPDUMP loadmodule "sipdump.so" #!endif #!ifdef WITH_DISPATCH loadmodule "dispatcher.so" #!endif # ----------------- setting module-specific parameters --------------- #!ifdef WITH_RTPENGINE # ----- rtpengine params -----+ modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:12222") #!endif #!ifdef WITH_TLS # ----- tls params -----+ modparam("tls", "xavp_cfg", "tls") modparam("tls", "config", "/etc/kamailio/tls.cfg") modparam("tls", "connection_timeout", 10) modparam("tls", "ssl_release_buffers", 1) modparam("tls", "send_close_notify", 1) modparam("tls", "session_cache", 0) #!endif #!ifdef WITH_SIPDUMP modparam("sipdump", "enable", 1) #!endif #!ifdef WITH_DISPATCH #---------- dispatch modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_interval", 300) #!endif # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/run/kamailio/kamailio_ctl") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { # per request initial checks route(REQINIT); #check who is the sender route(INITCHECK); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # update $du to set the destination address for proxying #$du = "sip:" + $rd + ":9"; route(RELAY); exit; } route[INITCHECK] { if(from_uri =~ ".*microsoft.com") { setflag(FROM_TEAMS); $du = "sip:" + "PBX_IP"; route(HANDLE_RTP_FROM_TEAMS); } else if(from_uri =~ ".*" + "PBX_IP") { setflag(FROM_PBX); $du="sip:sip.pstnhub.microsoft.com;transport=tls"; route(HANDLE_RTP_FROM_PBX); } else { exit; } } #Manage RTP & transcoding comming from Teams to PBX route[HANDLE_RTP_FROM_TEAMS] { if (has_body("application/sdp")) { t_on_reply("PBX_REPLY_TO_TEAMS"); rtpengine_manage("RTP codec-mask=all codec-transcode=PCMA replace-origin replace-session-connection ICE=remove"); record_route(); t_relay_to_udp("PBX_IP","5060"); } } #Manage RTP & transcoding comming from PBX to Teams route[HANDLE_RTP_FROM_PBX] { if (has_body("application/sdp")) { t_on_reply("TEAMS_REPLY_TO_PBX"); rtpengine_manage("SRTP codec-mask=all ICE=force codec-transcode=PCMA replace-origin replace-session-connection"); record_route_preset("SBC_FQDN:5061;transport=tls"); add_rr_param(";r2=on"); $rd = "sip.pstnhub.microsoft.com"; $td = "SBC_FQDN"; $fd = "SBC_FQDN"; #Set TLS SNI (server name & server id) $xavp(tls=>server_name) = "SBC_FQDN"; $xavp(tls=>server_id) = "SBC_FQDN"; t_relay(); } } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS")) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; #Teams reINVITEs if(isflagset(FROM_TEAMS)) { t_relay_to_udp("PBX_IP","5060"); exit; } # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails #set coresponding cert on transactions if($fd == "SBC_FQDN") { $xavp(tls=>server_name) = "SBC_FQDN"; $xavp(tls=>server_id) = "SBC_FQDN"; } } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); } #PBX On Reply onreply_route[PBX_REPLY_TO_TEAMS] { if (has_body("application/sdp")) { rtpengine_manage("SRTP codec-mask=all codec-transcode=PCMA replace-origin replace-session-connection media-address=PUBLIC_IP"); } } #From Teams On Reply onreply_route[TEAMS_REPLY_TO_PBX] { if (has_body("application/sdp")) { rtpengine_manage("RTP codec-mask=all codec-transcode=PCMA replace-origin replace-session-connection media-address=PUBLIC_IP"); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { if (t_is_canceled()) exit; } event_route[tm:local-request] { if(is_method("OPTIONS") && $ru =~ "pstnhub.microsoft.com") { append_hf("Contact: <sip:SBC_FQDN:5061;transport=tls>\r\n"); } xlog("L_INFO", "Sent out tm request: $mb\n"); }

6 5

Manipulating SDP IP for Inbound Calls
by Edward Romanenco 02 Aug '20

02 Aug '20

Hey guys, I am working on a project involving Kamailio dockerezation, which is meant to run alongside Freeswitch and RTPEngine containers, on the basis of a Docker-Compose file which is launched on top of a CentOS 7.7 host system. Anyway, I would love to know if there is any way to manipulate / mask the IP addresses that are being appended to a status 183 response for an incoming invite. For some reason which I am trying to figure out in parallel, Freeswitch uses the local network bridge subnet instead of the defined external RTP IPs, and I was wondering - Can I manipulate them using Kamailio? I know that Mangler module can do it for outbound calls, but can I do the same for inbound? v=0. o=FreeSWITCH 1595974788 1595974789 IN IP4 172.18.0.40. s=FreeSWITCH. c=IN IP4 172.18.0.40. t=0 0. m=audio 45878 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16 Edward

2 1

lua_dostring sr vs KSR
by Julien Chavanton 31 Jul '20

31 Jul '20

lua_dostring("sr.pv.seti(\"$$var(var1)\", $avp(var1) < $avp(var3) and 1 or 0)"); results in error : kamailio app lual_dostring(): error from Lua: attempt to index global 'sr' (a nil value) replacing sr with KSR is working : lua_dostring("KSR.pv.seti(\"$$var(var1)\", $avp(var1) < $avp(var3) and 1 or 0)"); Shall we update the docs for https://kamailio.org/docs/modules/devel/modules/app_lua.html#app_lua.f.lua_… ?

2 4

Issue with kamctl after upgrading to 5.3.4
by Igor Potjevlesh 31 Jul '20

31 Jul '20

Hello there! I'm having trouble with kamctl since I upgraded to 5.3.4. The command is working fine with root user but not with another user. The problem seems to be that the command try to access to /var/run/kamailio/kamailio_rpc.fifo even if the file exists. I guess it's a permission problem. But I was thinking that FIFO is no more used. Only jsonrpcs. How can I fix that? Regards, Igor. -- L'absence de virus dans ce courrier électronique a été vérifiée par le logiciel antivirus Avast. https://www.avast.com/antivirus

3 11

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sr-users July 2020