- sr-users - kamailio.org

Preprocessor ERROR! on upgrading to 5.4.0
by sagar malam 08 Aug '20

08 Aug '20

Hello , I was using Kamailio 5.3.3 . After upgrading to 5.4.0 i am getting below error with same cfg file which worked well with 5.3.3. 0(102523) ERROR: <core> [core/ppcfg.c:235]: pp_ifdef_level_error(): different number of preprocessor directives: 2 more #!if[n]def as #!endif What does this mean? Any clue to solve this? -- Thanks, Sagar

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In memory ISP Info
by Ilie Soltanici 07 Aug '20

07 Aug '20

Hello, What will be the best way to save ISP Information in Kamailio memory? I have the following information for each ISP into the DB: *ISP Name | ISP IP | ISP Username | ISP Password | ISP Prefix* ISP1 | 1.1.1.1 | user | pass | 1234 This info it's not changed/updated too often, so I want to avoid querying the DB on every call and get this info directly from the memory. I was thinking to use htable for that, so I created a htable as an array … [View More]

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Using wildcard certificates for Kamailio server
by Leonid Fainshtein 07 Aug '20

07 Aug '20

Hello, Is it permitted to use the wildcard TLS certificates for Kamailio server? In reality, it works (tested with v.5.4) but the RFC-5922 disables the wildcard certificates usage: "Implementations MUST match the values in their entirety: Implementations MUST NOT match suffixes. For example, "foo.example.com" does not match "example.com". Implementations MUST NOT match any form of wildcard, such as a leading "." or "*." with any other DNS label or … [View More]

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Deletion of Prometheus metrics
by Marat Gareev 07 Aug '20

07 Aug '20

Hello, I have confuse about this timeout: https://www.kamailio.org/docs/modules/devel/modules/xhttp_prom.html#xhttp_p… Is there reason for forced deletion of unused metrics? Let's imagine a situation: Zabbix reads metrics at a certain interval, unused metrics are deleted if they are not used for a period of time (xhttp_prom_timeout). I drew 2 cases when Zabbix will receive the same value, but we cannot say unambiguously whether these are new values or not. [image: xhttp_prom_timeout.jpg] Regards, Marat

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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($sel(cfg_get.… [View More]pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($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 [View Less]

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Info: 2 Kamailio presentations soon live at ClueCon - 1:30pm CST (20:30 Berlin)
by Daniel-Constantin Mierla 06 Aug '20

06 Aug '20

Hello, quick info for those that want to watch the 2 presentations about Kamailio at ClueCon Deconstructed, starting in about 1 hour 40 minutes from now (or 13:30CST, 20:30Berlin/Paris/Madrid/...). One will be done by me and the 2nd by Fred Posner. A few more details are available at: - https://www.kamailio.org/w/2020/07/cluecon-deconstructed-2020/ The sessions should be live streamed on youtube - should appear in the freeswitch channel there at the time of start: - … [View More]

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Audio on Calls Stopping After 15 Minutes
by admin＠conradit.com 06 Aug '20

06 Aug '20

All, hi: I'm a Kamailio newbie! I've managed to set up a proxy server with the minimal configuration file found here: https://github.com/kamailio/kamailio/blob/master/misc/examples/mixed/kamail…. The issue that I'm having is that the audio on calls simply stop precisely after 15 minutes. It's always precisely at 15 minutes. The calls continue connected but without any audio. A summary about my set up: 1. This is an Amazon AWS EC2 server so it doesn't have a public IP directly attached … [View More]

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Use-case does Kamailio support PBX - webrtc (web sockets)
by Johnny Ritzer 06 Aug '20

06 Aug '20

We have a PBX that doesn’t support websockets . But we want a PWA webaapp ctxSIP that uses sip.js to work . Due to its not SIP generic do we setup like this PBX <----> Kamailio <---->webapp Im looking for developers for android build via react native or ionic framework but not sure what solution/archtectiure Question. Does Kamailio register the SIP user/pass on the Kamailio server meaning it keeps it registered. And thus the ctxSIP client app just handles the pass thru for … [View More]

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Info: Kamailio Getting Started Workshop at ClueCon - Live in 1 hour
by Daniel-Constantin Mierla 05 Aug '20

05 Aug '20

Hello, in about 1 hour I will do a quick (1 hour long) workshop at ClueCon deconstructed. It can be watched live on youtube -- should appear in their youtube channel: * https://www.youtube.com/channel/UC0i2xbniNHJQNfPPs_v4T0Q Or if you want to join the live video conference, which allows you to ask questions and interact with the participants, use an webrtc capable browser and go to: * https://cluecon.cantina.video/ Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.… [View More]

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kamailio and rtpengine_manage() / rtpengine_delete()
by Karsten Horsmann 05 Aug '20

05 Aug '20

Hi list, I use rtpengine 7.2.1 branch with kamailio 5.2.x and 5.3.x in different setups. Due to some issues with my 5.2.x config I saw the "Total timed-out sessions via TIMEOUT" grow but "Total regular terminated sessions never" goes up from 0. That was an indicator for an non-existing rtpengine_manage() or rtpengine_delete() at the end of dialogs. Okay. Now I want to observe that a bit more. I use parallel forking to webrtc clients, that i guess will be the "more Offers" than Answers. Any … [View More]

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Redundancy of Progress Call using Kamailio
by Amit Pal 04 Aug '20

04 Aug '20

Hi Team , Hope all is fine. My requirement : I have a customer who wants high availability for ringing calls and calls that in process of being set up. Can you suggest an alternate mechanism to achieve replication of transaction state. Can you guide how it can be achieved in Kamailio or is there a mechanism to get commercial support to achieve the desired objective. I am waiting your response. Thanks Regards Amit Pal

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Kamailio Kubernetes AWS NLB UDP
by Ewgeny Berladin 04 Aug '20

04 Aug '20

Hi ! Does anyone have success Kamailio deployments via service on AWS EKS with UDP port exposing ? apiVersion: v1 kind: Service metadata: name: sip-proxy annotations: service.beta.kubernetes.io/aws-load-balancer-type: "nlb" spec: selector: app: kamailio-sip externalTrafficPolicy: Local type: LoadBalancer ports: - name: sip-up protocol: UDP port: 5060 As result we have this ERROR: ---- ------ ---- ---- -… [View More]

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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 … [View More]

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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 … [View More]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 [View Less]

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No more updates on Official Precise repositories
by Victor Seva 04 Aug '20

04 Aug '20

Hello, Ubuntu Precise is EOL since April 28, 2017 [0] and in order to support newer Ubuntu releases our deb build system will no longer produce newer packages for precise. Repositories will still be there but don't expect any updates. Cheers, Victor [0] https://wiki.ubuntu.com/Releases

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[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?

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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,

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topoh problem with bye from trunk
by mohsen khashei 04 Aug '20

04 Aug '20

Hi all. I use topoh module for topology hiding and this is my config for topoh: modparam("topoh", "mask_key", "my key") modparam("topoh", "mask_callid", 1) modparam("topoh", "callid_prefix", "!!:") modparam("topoh", "sanity_checks", 1) modparam("topoh", "mask_ip", "x.x.x.x") modparam("topoh", "uri_prefix_checks", 1) modparam("topoh", "uparam_name", "line") modparam("topoh", "uparam_prefix", "sr-") modparam("topoh", "vparam_name", "branch") modparam("topoh", "vparam_prefix", "z9hG4bKsr-"… [View More]

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Kamailio-IMS - Public Network Issue
by Pavithra Mohanraja 04 Aug '20

04 Aug '20

Hi, I have deployed kamailio-ims-5.3.5 as docker containers. I had private IP for containers. So when i try to establish a call from the public network , The via header is taken as container Private IP even if i use advertise address as public IP. So ACK is not reaching the pcscf from ue. Any idea can be provided for this issue. Kindly help. Thanks, Pavithra

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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 … [View More]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"); } [View Less]

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 … [View More]

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 … [View More]

3 11

publishing hash values for download files of releases
by Daniel-Constantin Mierla 30 Jul '20

30 Jul '20

Hello, being discussed during the last devel meetings, I published the md5, sha1 and sha256 hash values for the tarballs with sources and i386 binaries we make available for download on kamailio.org on each release -- e.g., for 5.4.0: * https://www.kamailio.org/pub/kamailio/5.4.0/src/ * https://www.kamailio.org/pub/kamailio/5.4.0/bin/ Before making a more official announcement about it and adding to the download/install docs, I want to discuss a little bit here and get to the … [View More]

2 2

Kamailio v5.4.0 Released - new major version is out
by Daniel-Constantin Mierla 29 Jul '20

29 Jul '20

Kamailio v5.4.0 is out – it comes with 5 new modules and a considerable set of improvements touching again more than 100 existing modules. You can read detailed release notes at: * https://www.kamailio.org/w/kamailio-v5-4-0-release-notes/ Many thanks to all developers and community members that made possible this release. v5.4.0 brings more flexibility and optimizations across many components, enhancements to load balancer, stateless active calls tracking, STIR/SHAKEN support, … [View More]

1 0

The plan for releasing Kamailio v5.4.0
by Daniel-Constantin Mierla 29 Jul '20

29 Jul '20

Hello, we are considering to release the next major stable release, respectively 5.4.0, on Wednesday, July 29, 2020. The branch 5.4 was already created, several doc resources were already added (wiki pages with what is new, core/variables/transformations/rpcs/stats cookbooks, alphabetic indexes, ...), at least debian/ubuntu packages should be built nightly, ... If anyone is testing an upgrade from branch 5.3 to 5.4 and encounters changes that should be done in Kamailio config, add notes … [View More]

1 2

PHONENUM Module on Centos7
by Duarte Rocha 28 Jul '20

28 Jul '20

Greetings, I've trying to use the module PHONENUM on my Kamailio on CentOS 7. However, getting the required libphonenumber library hasn't been possible. Is there any CentOs7 repository for this ? What's the best way to work with this module on CentOS ? Best Regards,

2 1

Kamailio IMS with custom AVPs
by Gaston Siffert 28 Jul '20

28 Jul '20

Hello all, I am investigating on how to integrate few IMS modules in the Platform I am working on. At the moment the main drawback for me is to find a way to export data from the Diameter Answers. In the RFC 7155, it’s written that the AA-Answer could have a list of customisable fields. While looking at the implementation of the Rx_AAR (IMS QOS) in Kamailio, the module does not seems to export any of the fields from the AA-Answer except the return code. The same applies for the implementation … [View More]

1 0

PHONENUM Module on Centos7
by Ilie Soltanici 28 Jul '20

28 Jul '20

Hello, If you want to use c++ library, then you can try to compile it by yourself: yum install cmake boost boost-devel boost-doc gtest gtest-devel protobuf protobuf-devel libicu libicu-devel $ cd libphonenumber/cpp $ mkdir build $ cd build $ cmake .. $ make $ ./libphonenumber_test cp -r libphonenumber.so* /usr/lib64/ cp -r libgeocoding.so* /usr/lib64/

1 0

ERROR: kazoo [kz_pua.c:131]: kz_pua_update_presentity(): in start_transaction
by Denys Pozniak 28 Jul '20

28 Jul '20

Hello! Sometimes presence server stops sending SIP NOTIFY events with syslog output below (after Kamailio restarting everything is restored): Jul 27 17:07:28 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: db_mysql [km_dbase.c:602]: db_mysql_start_transaction(): transaction already started Jul 27 17:07:28 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: kazoo [kz_pua.c:131]: kz_pua_update_presentity(): in start_transaction Jul 27 17:07:29 prd-sip-presence-2 /usr/sbin/kamailio[4838]: … [View More]ERROR: db_mysql [km_dbase.c:602]: db_mysql_start_transaction(): transaction already started Jul 27 17:07:29 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: kazoo [kz_pua.c:131]: kz_pua_update_presentity(): in start_transaction Jul 27 17:07:37 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: db_mysql [km_dbase.c:602]: db_mysql_start_transaction(): transaction already started Jul 27 17:07:37 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: kazoo [kz_pua.c:131]: kz_pua_update_presentity(): in start_transaction Jul 27 17:07:37 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: db_mysql [km_dbase.c:602]: db_mysql_start_transaction(): transaction already started Jul 27 17:07:37 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: kazoo [kz_pua.c:131]: kz_pua_update_presentity(): in start_transaction Jul 27 17:07:41 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: db_mysql [km_dbase.c:602]: db_mysql_start_transaction(): transaction already started Jul 27 17:07:41 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: kazoo [kz_pua.c:131]: kz_pua_update_presentity(): in start_transaction Jul 27 17:07:41 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: db_mysql [km_dbase.c:602]: db_mysql_start_transaction(): transaction already started Jul 27 17:07:41 prd-sip-presence-2 /usr/sbin/kamailio[4838]: ERROR: kazoo [kz_pua.c:131]: kz_pua_update_presentity(): in start_transaction Where is a possible source of the problem? Kamailio 5.1.2 with MySQL modparam("presence", "db_url", DBURL) modparam("presence", "subs_db_mode", 3) modparam("presence", "timeout_rm_subs", 0) modparam("presence", "expires_offset", 0) modparam("presence", "max_expires", 1800) modparam("presence", "db_update_period", 30) modparam("presence", "clean_period", 180) modparam("presence", "send_fast_notify", 1) modparam("presence", "pres_htable_size", 32) modparam("presence", "subs_htable_size", 32) modparam("presence", "publ_cache", 0) modparam("presence", "notifier_processes", 0) modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) modparam("presence_xml", "integrated_xcap_server", 0) modparam("presence_xml", "disable_presence", 1) modparam("presence_xml", "disable_winfo", 1) modparam("presence_dialoginfo", "force_single_dialog", 1) modparam("presence_dialoginfo", "force_dummy_dialog", 1) modparam("dialog", "dlg_flag", 4) modparam("dialog", "db_mode", 0) modparam("dialog", "dlg_match_mode", 1) modparam("kazoo", "node_hostname", NODE_HOSTNAME) modparam("kazoo", "amqp_connection", "AMQP_URL") modparam("kazoo", "pua_mode", 1) modparam("kazoo", "db_url", DBURL) modparam("kazoo", "amqp_consumer_processes", 10) modparam("kazoo", "amqp_max_channels", 50) -- BR, Denys Pozniak [View Less]

2 1

Kamailio CentOS 7 repos broken
by Patrick Wakano 28 Jul '20

28 Jul '20

Hi list, Hope you are all good! I've been trying to install kamailio 5.2.7 from the centos 7 repo, but it is failing in both repos..... With the http://rpm.kamailio.org/centos/ repo it fails with: Error: Package: kamailio-sipcapture-daemon-config-5.2.7-0.el7.centos.x86_64 (kamailio-5.2) Requires: kamailio-sipcapture = 5.2.7 With the http://download.opensuse.org/repositories/home:/kamailio:/v5.2.x-rpms/CentO… repo it fails with: kamailio-5.2.7-7.el7.centos.x8 FAILED =============… [View More]

2 1

Registrar - registered match_received
by Ilie Soltanici 27 Jul '20

27 Jul '20

Hello, Yes, i have it there but the variable name was mismatching with the module variable name: $xavp(regcfg=>match_received) = $su; modparam("registrar", "xavp_cfg", "reg") It's all good now. Thanks for your help.

1 0

Registrar - registered match_received
by Ilie Soltanici 27 Jul '20

27 Jul '20

Hello, Trying to check if an extension it's already registered with the same received path. if (registered("location","$fu",2)) { sl_send_reply("200","OK"); exit; }; But, it's not working for me, getting true even if the received path it's different: This is what I'm getting from Kamailio: [root@dev ilie.soltanici]# /usr/local/sbin/kamcmd ul.dump { Domains: { Domain: { Domain: location Size: 1024 … [View More] AoRs: { Info: { AoR: 101 HashID: -1698832128 Contacts: { Contact: { Address: sip:101@192.168.1.10:12767;ob Expires: 2452 Q: 1.000000 Call-ID: 3fa6c9f7db404e8e93a9664c3634ddf7 CSeq: 31114 User-Agent: MicroSIP/3.19.31 Received: sip: 192.168.1.10:12767 Path: [not set] State: CS_SYNC Flags: 0 CFlags: 12288 Socket: udp: 192.168.1.1:5060 Methods: 8159 Ruid: uloc-b-5f1eea5d-57b0-1 Instance: [not set] Reg-Id: 0 Server-Id: 11 Tcpconn-Id: -1 Keepalive: 1 Last-Keepalive: 1595862742 Last-Modified: 1595861637 } } This is the SIP Trace: 2020/07/27 16:17:12.214151 192.168.1.10:12610 -> 192.168.1.1:5060 OPTIONS sip:dev.local SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK-91-1-0 From: "Test Account" <sip:101@dev.local>;tag=91SIPpTag001 To: sip:dev.local Call-ID: 1-91(a)192.168.1.10 User-Agent: SIPP CSeq: 1 OPTIONS Content-Length: 0 As you can see the source port it's different 12767/12610, but the registered function is still getting true. Thank you. [View Less]

2 1

Install 5.3 on Ubuntu 20.04 With MySQL 8
by Mark Boyce 27 Jul '20

27 Jul '20

Hi all This is installing 5.3 for Bionic (18.04) on Focal (20.04) but as I know you’re in the throws of building 5.4 thought I'd throw this out there in case it changes anything; Installing kamailio from; deb http://deb.kamailio.org/kamailio53 bionic main The following packages have unmet dependencies: kamailio : Depends: libreadline7 (>= 6.0) but it is not installable kamailio-mysql-modules : Depends: libmysqlclient20 (>= 5.7.11) but it is not installable kamailio-redis-… [View More]

2 6

Using GeoIP2 AS Lookup?
by Mark Boyce 27 Jul '20

27 Jul '20

Hi all Just looking at the latest GeoIP2 MaxMind databases (now requires registration, but still free) and noticed that they also include the AS (ISP) lookup one in the free offering. Wondering if this is another way to facilitate better security for users on dynamic IP. Typically working from home these days. So, rather than just limiting an end device to a country we could limit it to a particular ISP within that country. Has anyone tried this? Have I missed a reason why this wouldn’t … [View More]

2 5

kamailio 5.3.5 - none local route header
by Jonathan Hunter 27 Jul '20

27 Jul '20

Hi Guys, Sorry if this is too generic or silly a question but I am testing kamailio 5.3.5 within Azure and its behind NAT. I am having an issue with passing an ACK from our internal side back out to the carrier side and it appears having run loose_route a none local route-header is consistently being removed by kamailio at this point. ie Route: <sip:CARRIER_IP;lr> Is this normal or configuration related? As I am unsure within what scenario I would remove a route header which doesnt … [View More]

2 3

Redis error: Server closed the connection / nbd_redis / kamailio 5.3.5
by Karsten Horsmann 27 Jul '20

27 Jul '20

Hi everyone, I get some strange errors from nbd_redis but my setup is working. kamailio 5.3.5 on CentOS 7.7.1908. Redis-Server is an haproxy frontend that pushed the stuff to an redis-sentinel cluster (with and really old version redis-3.2.1-2.el7.remi.x86_64). Any ideas on that? Jul 22 13:42:52 siptrunk5 kamailio[112377]: ERROR: ndb_redis [redis_client.c:1083]: redisc_exec_argv(): Redis error: Server closed the connection Jul 22 13:43:34 siptrunk5 kamailio[112359]: ERROR: ndb_redis [… [View More]

3 2

failed to lock the memory pages (disable swap) WARNING
by Joel Serrano 27 Jul '20

27 Jul '20

Hi all, Can anyone give me some info on what this warning means: Jul 7 17:02:29 csbc02 csbc[16006]: WARNING: <core> [core/daemonize.c:596]: mem_lock_pages(): failed to lock the memory pages (disable swap): Cannot allocate memory [12] Yes, if I set mlock_pages=no I don't see the warning, but I'm trying to understand why can't it allocate with mlock_pages=yes. Joel.

4 4

Kamailio Installation Steps
by Abdul Rasheed 26 Jul '20

26 Jul '20

Hello All , I am trying to setup Kamailio on Debian . May i know which document should be followed for installing the latest version of Kamailio on Debian 10 Buster. ? Thanks and Regards Abdul Rasheed

2 2

Dialplan: using $avp("key") on subst rule with regexp capturing group
by João Vitor Arruda 24 Jul '20

24 Jul '20

Hello, I'm trying to use the dialplan with $avp("key") on match_exp and subst_exp but i'm getting an error message when there is a regex capturing group on subst_exp to be used on repl_exp This is the config used: modparam("dialplan", "match_dynamic", 1) $avp(s:PSTNAC) = "0"; Dialplan entry: dpid = 1 match_op = 1 match_exp = ^$avp(s:PSTNAC) match_len = 0 subst_exp = ^$avp(s:PSTNAC)(.*) repl_exp = 9\1 After executing dp_translate while the dialplan is properly matched the following … [View More]

1 2

how to achive destination number based routing before falling back to dispatcher-module
by Christoph Russow 24 Jul '20

24 Jul '20

Hi all, im fairly new to Kamailio and currently setting up our new sip-server systems so, hello everyone! right now i have a Kamailio SBC which routes all calls using the dispatcher module to our internally setup application servers (freeswitch). i just got the idea that we might want specific routing based on destination phone number in addition to simple loadbalancing. can anyone point me in the right direction where i should look for something i can place before the dispatcher … [View More]

4 9

HEP siptrace HEPv3 truncated and ignored by heplify-server (Homer 7)
by José Lopes 24 Jul '20

24 Jul '20

Hello all, I am using the siptrace module on Kamailio 5.3.5 and I am trying to monitor the SIP over websockets traffic with Homer 7.8.1 (heplify-server). I notice a strange behavior, some packets are not presented on Homer. I checked heplify-server and i notice the next warning: heplify-server: 2020/07/22 18:41:09.425513 decoder.go:103: WARN HEP packet length is 8192 but should be 11663 So I think the packets are truncated by 8192 (packet size), so they are ignored by heplify-server. I have … [View More]

2 2

new in upcoming Kamailio 5.4.0
by Henning Westerholt 23 Jul '20

23 Jul '20

Hello, I've updated the "New in version 5.4.0" document in our wiki. It is again an impressive list! * In this development period (started October 2019) almost 1500 commits have been added to the repository * 5 new modules have been added (pv_headers, kafka, secsipid, systemdops, dlgs) * Over 70 modules have been extended, many of them substantially (dispatcher, pv, tls, presence etc..) * Countless bugfixes and documentation extensions/fixes have been added as well * Many … [View More]

5 15

TAS recommendation
by BALL SUN 23 Jul '20

23 Jul '20

Hi We recently setup Kamailio as the IMS core in our test lab, and we would like to test scenarios with TAS, is there any recommendation? Does Kamailio able to setup as a SIP AS? if yes, any online instruction on how we can make it works? Thanks RBK

1 0

Dispatcher does not go to 2nd server
by Aristeidis Tsitras 22 Jul '20

22 Jul '20

I have a Kamailio and 2 asterisk servers. All users are created in both of the asterisk servers. I am forwarding the registration to asterisk. The problem is that it is always used on only one server from the list. Even if one goes to shutdown, then there is not any registration sent to the available server. Even if *some *of the extensions can be seen registered in both of the asterisk's, if the secondary goes down, then there are no services for the phones. I am attaching the kamailio.cfg. … [View More]

6 12

Siremis - PassThru Authentication
by Adam - UC-Reseller 21 Jul '20

21 Jul '20

I have recently installed Siremis And have set up a domain pointing to it, how do I configure it so registration requests to this domain are passed thru to the actual PBX Adam

2 1

P-Asserted-Identity in SIP INVITE from Kamailio IMS to PSTN (MGCF)
by BALL SUN 21 Jul '20

21 Jul '20

Hi We have a question. We are preparing a setup to bridge the SIP call from Kamailio IMS to PSTN, but we are puzzling how we can present the P-Asserted-Identity in the SIP INVITE. By default, the PAI is in sip:user@domain, but we believed we should use<tel> format in the PAI when bridge to PSTN. question is 1. should we have two PAI headers?, or 2. should we paste <tel> format in PAI User Part and domain in PAI Host Part?, or 3. Should we only use one PAI header with tel format? … [View More]

1 1

Kamailio load balance to PBXes
by John Tuxies 20 Jul '20

20 Jul '20

Trying to setup a lab where i would like tohave: -kamailio (5.3 from deb repo) in front to load balance. Keep registrations. IP: 192.168.1.100 -Asterisk _1 (16 LTS). IP: 192.169.1.101.It <http://192.169.1.232.It> will serve as media server for IVR, VM, etc. Bind port is 5080. A sip trunk to Kamailio in sip.conf -Asterisk-2 (16 LTS). IP: 192.168.1.102. Identical config to Asterisk-1. Tried to follow this https://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb, but it … [View More]

2 2

Kamailio load balance to PBXes
by John Tuxies 20 Jul '20

20 Jul '20

Trying to setup a lab where i would have: -kamailio in front to load balance. Keep registrations. IP: 192.168.1.100 -Asterisk _1 (16 LTS). IP: 192.169.1.101.It <http://192.169.1.232.It> will serve as media server for IVR, VM, etc. Bind port is 5080. A sip trunk to Kamailio in sip.conf -Asterisk-2 (16. IP: 192.168.1.102. Identical config to Asterisk-1. Tried to follow this https://kb.asipto.com/asterisk:realtime:kamailio-3. 0.x-asterisk-1.6.2-astdb, but it is quite old and it does jot … [View More]

1 0

Kamailio load balance to PBXes
by John Tuxies 20 Jul '20

20 Jul '20

Trying to setup a lab where i would have: -kamailio in front to load balance. Keep registrations. IP: 192.168.1.218 -Asterisk _1. IP: 192.169.1.232.It will serve as media server for IVR, VM, etc -Asterisk-2. IP: 192.168.1.233. Identical config to Asterisk-1. Tried to follow this https://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb, but it is quite old and it does jot apply to newer versions of Asterisk and Kamailio. Up to jow i have the I would like advice on setup and configuration,

1 0

Kamailio & Freeswitch as media and SBC
by John Tuxies 20 Jul '20

20 Jul '20

Trying to setup Kamailio with Freeswitch in a Debian machine as in the http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc, but Kamailio fails to start and gives a ton of errors. Before of the alterations Kamailio and Siremis start correctly and i can easily create users through Siremis. After the changes to include Freeswitch's blocks, it fails to start. All run in the same machine with IP 192.168.1.213. I am attaching kamailio.cfg (named as test1.txt) if someone could spot the … [View More]

3 8

SIP Security handling in Kamailio IMS
by BALL SUN 20 Jul '20

20 Jul '20

Hi During the testing with Kamailio IMS, we found that SUBSCRIBE and NOTIFY message flow between UE and P-CSCF is shown below. SUBSCRIBE and its response: 192.168.1.102:9101 (port-c) -> 192.168.2.66:6101 (port-s) NOTIFY: 192.168.2.66:5101 (port-c) -> 192.168.1.102:9100 (port-s) based on the sip security, is it the expected behavior that they are using port-c and port-s? The reason why we asked is because we found that the NOTIFY did not properly propagate to the UE due to different … [View More]

2 2

the list with sip compact header names
by Daniel-Constantin Mierla 17 Jul '20

17 Jul '20

Hello, anyone knowing by heart or a place listing all compact header names? I found: - https://www.cs.columbia.edu/sip/compact.html but it says: Last updated 01/10/2008 Maybe newer RFC introduced other compact header names. Or, in case you are aware of some other compact names not listed there, just say it here, as I am working a bit on updating the parser on this scope. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/… [View More]

2 2

CommCon 2020 Online - Now presentation about Kamailio
by Daniel-Constantin Mierla 16 Jul '20

16 Jul '20

Hello, being a bit late with this announcement, as it started a couple of minutes ago, but if anyone wants to watch the live presentation about Kamailio by Fred Posner at CommCon 2020 Online, here is the link: - https://www.youtube.com/watch?v=FGZF7dYdRng Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 0

Reset Prometheus counter once
by Marat Gareev 16 Jul '20

16 Jul '20

Hello. Are there any ways to reset the Prometheus counter once? I can do something like this, but counter will be reset on each request loadmodule "xhttp_prom.so" > modparam("xhttp_prom", "prom_counter", "name=mycounter;") > route { > prom_counter_reset("mycounter"); > } Otherwise, I will not receive the counter until a special and rare event occurs. Regards, Marat

3 4

RPC Socket stops answering after a few time.
by David VILLAUME 16 Jul '20

16 Jul '20

Hello, I’m having problem with RPC socket that stops answering after approximatively one or two hours. Version is 5.2.7 My principal host with exact same version and configuration has no problem, only the backup node (Keepalived + DMQ) encounters the problem. I could catch this log about the problem, but didn’t find any reason for this : Jun 18 10:53:33 sbc-kam-partner-b kamailio[83170]: ERROR: ctl [binrpc_run.c:600]: rpc_send(): ERROR: binrprc: rpc_send: send failed My configuration … [View More]

2 13

Routing specific number from kamailio to asterisk
by Hendra 834 16 Jul '20

16 Jul '20

Hi, Im new in VoIP, especially Kamailio. I want to connect my kamailio with asterisk by routing the call coming from kamailio to asterisk for specific number, also from asterisk to kamailio. For example, if kamailio client call ext number 0123, the will be routed to asterisk, if kamailio client call 0233 the call will not be routed to external proxy. And also how can I receive call form asterisk to kamailio ? Is there any real example, ? because I dont understand what to change in documentation example. Thanks

2 1

Is it possible to load both dialog and ims_dialog modules in the same configuration?
by Sumesh Soman 16 Jul '20

16 Jul '20

Hi, Is it possible to load and use both dialog and ims_dialog modules in the same server ? Even though ims_dialog docs says ims_dialog extends dialog module, I do see that some of the dialog module features are not present in ims_dialog. Example, wait_ack , track_cseq_updates are not present in ims_dialog. I have requirements to use ims_dialog mainly for ims_charging module and dialog module with wait_ack and track_cseq_updates parameters are enabled. -- Regards, Sumesh S.

1 0

5.4 compiler warning
by Juha Heinanen 14 Jul '20

14 Jul '20

Got this when building 5.4; CC (gcc) [sip-proxy] core/dset.o core/dset.c: In function 'uri_trim_rcv_alias': core/dset.c:1042:6: warning: variable 'proto' set but not used [-Wunused-but-set-variable] int proto; ^~~~~ -- Juha

2 2

Re: [SR-Users] xHTTP_PROM with older Kamailio
by Marat Gareev 13 Jul '20

13 Jul '20

Hello, Karsten, Thanks for the link. I will definitely examine it.

1 0

No of registartions
by Pavithra Mohanraja 13 Jul '20

13 Jul '20

Hi, Is there any way to query number of registrations happened in kamailio IMS. Kindly reply . Thanks, Pavithra

2 1

subsequent ds_select_dst with different groups
by David VILLAUME 13 Jul '20

13 Jul '20

Hello, Is it possible to use the following logic : Route() ds_select_dst (1,2) failure_route(1) ds_next_dst(first_dispatcher_ secondgateway) failure_route(newdispatcher_group) ds_select_dst (2,2) failure_route(secondispatcher_secondgateway) ds_next_dst() Thanks in advance. Regards, David

4 5

Compact SIP
by Abdoul Osséni 13 Jul '20

13 Jul '20

Hello, I use Kamailio and Asterisk. SIP Compact header is activated by ASTERISK ( https://www.cs.columbia.edu/sip/compact.html). I am trying to disable the SIP Compact header on KAMAILIO. Is it possible? Regards Abdoul OSSENI

3 4

Re: [SR-Users] xHTTP_PROM with older Kamailio
by Marat Gareev 12 Jul '20

12 Jul '20

I'm sorry. I mean *xhttp_prom.c* file instead of *xhttp_mod.c*. >

1 0

xHTTP_PROM with older Kamailio
by Marat Gareev 11 Jul '20

11 Jul '20

Hello, I have a customized Kamailio (kamailio 5.1.1 (x86_64/linux)), xHTTP_PROM <https://www.kamailio.org/docs/modules/devel/modules/xhttp_prom.html> was released in the 5.3.0 <https://www.kamailio.org/w/kamailio-v5-3-0-release-notes/> version. I'm trying to import the latest version of xhttp_prom to older Kamailio, but something goes wrong. This is simple config for test: log_stderror=yes > listen=5060 > debug=2 > mpath="/home/devel/build_dir/build/lib64/kamailio/… [View More]modules" > loadmodule "tm.so" > loadmodule "sl.so" > loadmodule "xlog.so" > loadmodule "pv.so" > disable_tcp=no > tcp_accept_no_cl=yes > loadmodule "xhttp.so" > loadmodule "xhttp_prom.so" > modparam("xhttp_prom", "xhttp_prom_stats", "all") > route { > sl_send_reply("200", "OK"); > exit; > } > event_route[xhttp:request] { > if (prom_check_uri()) { > prom_dispatch(); > } else { > xhttp_reply("404", "Not Found", "text/plain", "Wrong URL $hu\n"); > } > } then I send HTTP request $ curl http://localhost:5060/metrics but get error: > 6(110968) CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 11 > 0(110962) ALERT: <core> [main.c:746]: handle_sigs(): child process 110967 > exited by a signal 11 > 0(110962) ALERT: <core> [main.c:749]: handle_sigs(): core was generated > 0(110962) INFO: <core> [main.c:771]: handle_sigs(): terminating due to > SIGCHLD note: for http://localhost:5060/test request I get the expected message "Wrong URL /test". Backtrace for killed process > (gdb) c > Continuing. > Program received signal SIGSEGV, Segmentation fault. > 0x00007fefff3ff096 in atomic_cmpxchg_int (var=0x0, old=0, new_v=1) at > ../../core/mem/../atomic/atomic_x86.h:224 > 224 ATOMIC_FUNC_CMPXCHG(cmpxchg, "cmpxchgl %2, %1", int , int) > (gdb) bt > #0 0x00007fefff3ff096 in atomic_cmpxchg_int (var=0x0, old=0, new_v=1) at > ../../core/mem/../atomic/atomic_x86.h:224 > #1 0x00007fefff3ff0e2 in futex_get (lock=0x0) at > ../../core/mem/../futexlock.h:99 > #2 0x00007fefff4130bf in prom_metric_list_print (ctx=0x7fefff628da0 > <ctx>) at prom_metric.c:1319 > #3 0x00007fefff416468 in prom_stats_get (ctx=0x7fefff628da0 <ctx>, > stat=0x7fefff6281f0 <xhttp_prom_stats>) at prom.c:178 > #4 0x00007fefff3d4d10 in prom_send (ctx=0x7fefff628da0 <ctx>) at > xhttp_prom.c:361 > #5 0x00007fefff3d5faf in ki_xhttp_prom_dispatch (msg=0x7ffecb730ee0) at > xhttp_prom.c:418 > #6 0x00007fefff3d5feb in w_prom_dispatch (msg=0x7ffecb730ee0) at > xhttp_prom.c:428 > #7 0x0000000000461dbe in do_action (h=0x7ffecb730e00, a=0x7ff0007c7b78, > msg=0x7ffecb730ee0) at core/action.c:1067 > #8 0x000000000046e571 in run_actions (h=0x7ffecb730e00, a=0x7ff0007c7b78, > msg=0x7ffecb730ee0) at core/action.c:1565 > #9 0x0000000000461d2d in do_action (h=0x7ffecb730e00, a=0x7ff0007ca628, > msg=0x7ffecb730ee0) at core/action.c:1058 > #10 0x000000000046e571 in run_actions (h=0x7ffecb730e00, a=0x7ff0007c76f8, > msg=0x7ffecb730ee0) at core/action.c:1565 > #11 0x00007fefff62f649 in xhttp_process_request (orig_msg=0x7ff0007ebbd8, > new_buf=0x7ff0007cc280 "GET /metrics HTTP/1.1\r\nVia: SIP/2.0/TCP > 127.0.0.1:37000\r\nUser-Agent: curl/7.29.0\r\nHost: > localhost:5060\r\nAccept: */*\r\n\r\n", new_len=119) at xhttp_mod.c:298 > #12 0x00007fefff630d78 in xhttp_handler (msg=0x7ff0007ebbd8) at > xhttp_mod.c:385 > #13 0x0000000000520141 in nonsip_msg_run_hooks (msg=0x7ff0007ebbd8) at > core/nonsip_hooks.c:111 > #14 0x000000000058cef1 in receive_msg (buf=0x7feffee65bb0 "GET /metrics > HTTP/1.1\r\nUser-Agent: curl/7.29.0\r\nHost: localhost:5060\r\nAccept: > */*\r\n\r\n", len=85, rcv_info=0x7feffee658d0) > at core/receive.c:216 > #15 0x000000000061233f in receive_tcp_msg (tcpbuf=0x7feffee65bb0 "GET > /metrics HTTP/1.1\r\nUser-Agent: curl/7.29.0\r\nHost: > localhost:5060\r\nAccept: */*\r\n\r\n", len=85, rcv_info=0x7feffee658d0, > con=0x7feffee658b8) at core/tcp_read.c:1379 > #16 0x000000000061479e in tcp_read_req (con=0x7feffee658b8, > bytes_read=0x7ffecb731c7c, read_flags=0x7ffecb731c78) at > core/tcp_read.c:1611 > #17 0x00000000006174e9 in handle_io (fm=0x7ff0007d9720, events=1, idx=-1) > at core/tcp_read.c:1785 > #18 0x00000000006069b4 in io_wait_loop_epoll (h=0xaa76c0 <io_w>, t=2, > repeat=0) at core/io_wait.h:1065 > #19 0x00000000006194ff in tcp_receive_loop (unix_sock=14) at > core/tcp_read.c:1955 > #20 0x00000000004f1a43 in tcp_init_children () at core/tcp_main.c:4853 > #21 0x0000000000423f99 in main_loop () at main.c:1716 > #22 0x000000000042a5ee in main (argc=9, argv=0x7ffecb7323b8) at main.c:2646 > (gdb) > (gdb) p 'prom_metric.c'::prom_lock > $4 = (gen_lock_t *) 0x0 If I understood correctly, the problem is in an uninitialized variable static gen_lock_t *prom_lock = NULL; /* Lock to protect Prometheus metrics. */ So, is it possible to import xHTTP_PROM to older Kamailio? I will be glad for any advice. Thanks. [View Less]

1 0

loose_route failing with topos ?
by Mark Boyce 10 Jul '20

10 Jul '20

Hi all Just enabled topos with redis backend (topos/topos_redis/ndb_redis) The interesting bits of the calls are going from kamailio (5.3.5) to Asterisk with topos enabled for that leg All seems to be ok until the far end (Asterisk) sends a BYE. At which point kamailo passes message to WITHINDLG, in loose_route comes up false (-1) and the BYE gets missed, falling through to the “404 Not here” at the end of WITHINDLG. I’m guessing that loose_route should be able to “see inside” topos, or … [View More]

3 4

rtcp-mux is missing in the OK for WebRTC-Calls
by Benjamin Flügel | vio:networks 10 Jul '20

10 Jul '20

Hey guys, I'm trying to configure a Kamailio to work with a browser softphone based on SIPJS using WebRTC. So far it works great on Firefox but have a specific problem with chrome, when I want to make call from the softphone to another extension. After anwsering the call Chrome/the softphone sends a BYE immediately, because this line "a=rtcp-mux" is missing in the OK. The Kamailio is a proxy. Behind the Kamailio there is an Asterisk, which is responsible for the pbx-features. Those are my … [View More]rtpengine Flags for the Invite: rtpengine_manage: replace-origin replace-session-connection trust-address via-branch=extra rtcp-mux-demux DTLS=off SDES-on ICE=remove RTP/AVP And those are the flags for the response, in this case the OK: rtpengine_manage: replace-origin replace-session-connection rtcp-mux-offer rtcp-mux-accept generate-mid DTLS=off SDES-on ICE=force RTP/SAVPF direction=internal direction=external loop-protect It seems that the Kamailio ignores ths "rtcp-mux-offer rtcp-mux-accept" in the response. Can you help me get it to work? Here is the SIP-Dialog. The call is from extension 201 to the extension 2. INVITE: INVITE sip:2@mydomain SIP/2.0 Via: SIP/2.0/WSS g4n0lpfirgpv.invalid;branch=z9hG4bK5104892 To: <sip:2@mydomain.com> From: <sip:201@mydomain.com>;tag=aqplr71k05 CSeq: 2 INVITE Call-ID: sde09v49f8gqtt59oddn Max-Forwards: 70 Proxy-Authorization: Digest algorithm=MD5, username="201", realm="mydomain.com", nonce="sfdsdfdsfsdfdsfsdfaseqww", uri="sip:2@mydomain.com", response="81rewtrega23423r" Contact: <sip:kf6iduon@g4n0lpfirgpv.invalid;transport=ws;ob> Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: outbound User-Agent: SIPJS Content-Type: application/sdp Content-Length: 1916 v=0 o=- 5826889093965459811 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE 0 a=msid-semantic: WMS Jk5GQwzPaVSTxyIZER7RBqkkMNWCdmKmMdyO m=audio 54274 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126 c=IN IP4 mycomputerIP a=rtcp:9 IN IP4 0.0.0.0 a=candidate:1577908739 1 udp 2113937151 efd6d297-d186-4c07-87d8-933e5846a82d.local 54274 typ host generation 0 network-cost 999 a=candidate:842163049 1 udp 1677729535 mycomputerIP 54274 typ srflx raddr 0.0.0.0 rport 0 generation 0 network-cost 999 a=ice-ufrag:Ktxl a=ice-pwd:x5xDUb41GeOXAcHNQlra4yUN a=ice-options:trickle a=fingerprint:sha-256 2C:7C:32:AF:34:A3:9D:AE:C7:FD:92:68:DD:D8:AB:82:DB:F0:32:51:14:97:20:60:66:5C:2F:CF:B7:98:B8:A8 a=setup:actpass a=mid:0 a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id a=sendrecv a=msid:Jk5GQwzPaVSTxyIZER7RBqkkMNWCdmKmMdyO dc187be3-4cd0-4129-98e4-f804cd8d2c94 a=rtcp-mux a=rtpmap:111 opus/48000/2 a=rtcp-fb:111 transport-cc a=fmtp:111 minptime=10;useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:110 telephone-event/48000 a=rtpmap:112 telephone-event/32000 a=rtpmap:113 telephone-event/16000 a=rtpmap:126 telephone-event/8000 a=ssrc:2376769177 cname:RgPkYyxDrEJTpmO2 a=ssrc:2376769177 msid:Jk5GQwzPaVSTxyIZER7RBqkkMNWCdmKmMdyO dc187be3-4cd0-4129-98e4-f804cd8d2c94 a=ssrc:2376769177 mslabel:Jk5GQwzPaVSTxyIZER7RBqkkMNWCdmKmMdyO a=ssrc:2376769177 label:dc187be3-4cd0-4129-98e4-f804cd8d2c94 RINGING: SIP/2.0 180 Ringing Via: SIP/2.0/WSS g4n0lpfirgpv.invalid;rport=60196;received=mycomputerIP;branch=z9hG4bK5104892 Record-Route: <sip:internalIP;r2=on;lr=on;ftag=aqplr71k05;nat=yes;rtp=ws> Record-Route: <sip:externalIP;transport=ws;r2=on;lr=on;ftag=aqplr71k05;nat=yes;rtp=ws> From: <sip:201@mydomain.com>;tag=aqplr71k05 To: <sip:2@mydomain.com>;tag=as64fe0fc8 Call-ID: sde09v49f8gqtt59oddn CSeq: 2 INVITE Server: myserver Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <mycontact> P-Asserted-Identity: "Phone 2" <sip:2@mydomain.com> Content-Length: 0 OK: SIP/2.0 200 OK Via: SIP/2.0/WSS g4n0lpfirgpv.invalid;rport=60196;received=mycomputerIP;branch=z9hG4bK5104892 Record-Route: <sip:internal-ip;r2=on;lr=on;ftag=aqplr71k05;nat=yes;rtp=ws> Record-Route: <sip:external-ip;transport=ws;r2=on;lr=on;ftag=aqplr71k05;nat=yes;rtp=ws> From: <sip:201@mydomain.com>;tag=aqplr71k05 To: <sip:2@mydomain.com>;tag=as64fe0fc8 Call-ID: sde09v49f8gqtt59oddn CSeq: 2 INVITE Server: myserver Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <mycontact> P-Asserted-Identity: "Phone 2" <sip:2@mydomain.com> Content-Type: application/sdp Content-Length: 806 v=0 o=root 882840402 882840402 IN IP4 externalIP s=Asterisk PBX 11.22.0 c=IN IP4 externalIP t=0 0 a=rtpengine:4d633e3022f5 m=audio 16416 RTP/SAVPF 111 9 8 0 126 a=maxptime:60 a=mid:0 a=rtpmap:111 opus/48000/2 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:126 telephone-event/8000 a=fmtp:111 maxplaybackrate=16000; stereo=0; sprop-stereo=0; useinbandfec=0 a=fmtp:126 0-16 a=sendrecv a=rtcp:16417 a=setup:active a=fingerprint:sha-1 B1:67:4B:B8:47:89:E8:49:CD:DD:F8:FF:41:5C:83:72:D9:DE:4D:45 a=ptime:20 a=ice-ufrag:WBZN8m7c a=ice-pwd:mKVR6Jv0G0pvrTv7OfEm2wPW9Q a=ice-options:trickle a=candidate:3kddB3zBUV2n2jt0 1 UDP 2130706431 externalIP 16416 typ host a=candidate:3kddB3zBUV2n2jt0 2 UDP 2130706430 externalIP 16417 typ host a=end-of-candidates I hope, you can help me with this. If you have further question, I will try to answer them as best I can. Greetings, Ben [View Less]

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Development is open for future release of v5.5.x series
by Daniel-Constantin Mierla 10 Jul '20

10 Jul '20

Hello, the branch 5.4 was created, therefore the master branch is open for adding new features, to be part of future release series v5.5.x. Any bug fix committed to master that applies to 5.4.x or older stable branches should be backported as usual with "git cherry-pick -x ..." to appropriate branches like 5.4 or 5.3. Expect that v5.4.0 will be released in a few weeks from now. Based on the workflow used during the past years, the next future release v5.5.0 should be out after another 8-10 … [View More]

1 0

Branch 5.4 has been created
by Daniel-Constantin Mierla 10 Jul '20

10 Jul '20

Hello, the branch 5.4 has been created, to be used for releasing v5.4.x series. To check out this branch, the following commands can be used: git clone https://github.com/kamailio/kamailio kamailio-5.4 cd kamailio-5.34 git checkout -b 5.4 origin/5.4 Pushing commits in this branch: git push origin 5.4:5.4 Note that 5.4 is an official stable branch, so only bug fixes, missing kemi exports (discuss on sr-dev if not sure) or improvements to … [View More]

1 0

Drop local generated requests
by Patrick Wakano 10 Jul '20

10 Jul '20

Hello list, Hope you all doing fine! I have a case where I want to drop a local generated request but it looks like you can't drop the request from the tm:local-request route. I found this ticket https://github.com/kamailio/kamailio/issues/980, but I could not find somewhere in the docs or emails definitely saying that drop() is not supported from tm:local-request.... And in case we definitely can't drop the request in there, any ideas on how to avoid a local generated request of going out? I … [View More]

2 4

Help on dumping running config in Kamailio
by BALL SUN 09 Jul '20

09 Jul '20

Hi I need an urgent help, I accidentally pipe the output to the kamailio config, and now I lost the copy of my latest setting, is there a chance that I can dump the running config back to file? Thanks PLEASE HELP RBK

2 1

Branching 5.4 in git repository
by Daniel-Constantin Mierla 08 Jul '20

08 Jul '20

Hello, so far it looks pretty calm during the testing phase, some static analysis didn't reveal any new major issues (well, this is done also during development phase). Therefore, if nothing else is proposed or shows up, I am considering to create the git branch 5.4 next week on Thursday -- this branch will hold the code for 5.4.x release series. After branching 5.4, the master branch will be open again for new features. The 5.4.0 enters the release candidate phase, likely to have the first … [View More]

1 1

Maintenance work on kamailio.org server - Wed, July 8, 2020
by Daniel-Constantin Mierla 08 Jul '20

08 Jul '20

Hello, on Wednesday, July 8, 2020, there will be some work on Kamailio project infrastructure. Therefore there could be short intervals of downtime for online resources like the web server, wiki, documentation and downloads portals, mailing lists or the GIT repository mirror. Once the maintenance work is finished, we will post an update. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 2

30X redirect in interconnects - better alternative?
by Gerry | Rigatta.com 08 Jul '20

08 Jul '20

Hi, I would like to use Kamailio for load balancing incoming carrier traffic. We do currently IP authentication and call logic in Yate boxes. Ideally I would like to distribute calls with 30X redirects with the Kamailio dispatcher so that IP authentication and all logic can stay in the Yate boxes. However I have doubts that 30X redirects are generally accepted in interconnects. What is your experience with this? What is the possible alternative to redirects if one wants to keep IP … [View More]

3 2

Re: [SR-Users] sanity module checks fail for ACK with parameters in RURI - possible bug?
by George Diamantopoulos 08 Jul '20

08 Jul '20

Update: Disabling the topoh module on the proxy which produces the error seems to stop the failure from manifesting. I'll try using topos_redis instead, but should this be treated as a bug? BR, George On Wed, 8 Jul 2020 at 12:37, George Diamantopoulos <georgediam(a)gmail.com> wrote: > Hello again, > > Indeed $mb seems to contain garbage: > > SCRIPT_MB: ACK <BC><EA><8F> SIP/2.0#015#012Via: SIP/2.0/UDP > 172.30.154.189;TH=dcv;branch=z9hG4bK629b.… [View More]6af9302cd78dc58dffe817e60124f4ed.0#015#012Route: > <sip:RJ2U2c7mrFzQjgG5SSvyE8RVS9omgAA=@172.30.155.1 > ;lr;received=sip:2.2.2.2:32768 > ;ob;r2=on>,<sip:RJ2U2c7mrFzQjgG5SSvyE8RVS9omgAA=@3.3.3.3 > ;lr;received=sip:2.2.2.2:32768;ob;r2=on>#015#012Max-Forwards: > 68#015#012From: "Anonymous" <sip:unknown@voip.domain.com>;tag=as4bc9e324#015#012To: > <sip:voip-test-user-02@voip.domain.com>;tag=jw7z5s0zvc#015#012Call-ID: > 0138c6370346d1dd7f1b47f604b01f92(a)voip.domain.com#015#012CSeq > <http://0138c6370346d1dd7f1b47f604b01f92@voip.domain.com#015%23012CSeq>: > 102 ACK#015#012Content-Length: 0#015#012TH: dch#015#012#015#012 > > How can this be possible? Capturing traffic on wire shows the RURI I > pasted in my original message and there are no script operations on the > RURI before sanity_check() (message buffer above is printed just before > sanity_check() is run in REQINIT). > > BR, > George > > On Wed, 8 Jul 2020 at 11:18, George Diamantopoulos <georgediam(a)gmail.com> > wrote: > >> I'll post the message buffer ASAP, but in the meantime I don't see how >> config operations could affect the RURI. Here's everything that's happening >> until the sanity check involved: >> >> request_route { >> >> if(is_method("KDMQ")) { >> dmq_handle_message(); >> } >> >> # no connect for sending replies >> set_reply_no_connect(); >> >> if($ua =~ "friendly-scanner|sipcli|sipvicious|VaxSIPUserAgent") { >> # silent drop for scanners - uncomment next line if want to reply >> # sl_send_reply("200", "OK"); >> exit; >> } >> >> if (!mf_process_maxfwd_header("10")) { >> force_rport(); >> sl_send_reply("483","Too Many Hops"); >> exit; >> } >> >> # OPTIONS and NOTIFYs directed to myself >> if(is_method("OPTIONS|NOTIFY") && uri==myself && $rU==$null) { >> force_rport(); >> sl_send_reply("200","Keepalive"); >> exit; >> } >> >> # All keep-alive methods regardless of destination >> if ( $hdr(Event) == "keep-alive") { >> force_rport(); >> sl_send_reply("200","Keepalive"); >> exit; >> } >> >> if(!sanity_check("17895", "7")) { >> xlog("Malformed SIP request from $si:$sp\n"); >> exit; >> } >> >> BR, >> George >> >> On Wed, 8 Jul 2020 at 10:58, Daniel-Constantin Mierla <miconda(a)gmail.com> >> wrote: >> >>> Hello, >>> >>> check your config operations, because the R-URI seems to be the next >>> string (without quotes): "<BC><EA><8F>" >>> >>> You can try to print $mb in such case to see the entire SIP message >>> buffer. >>> >>> Cheers, >>> Daniel >>> On 08.07.20 09:48, George Diamantopoulos wrote: >>> >>> Hello Daniel, >>> >>> Thanks for the reply. Indeed there is, not sure how I managed to miss >>> that. And it wasn't about the schema after all: >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: DEBUG: {1 >>> <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity >>> [sanity.c:277]: check_ruri_scheme(): check_ruri_scheme entered >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: DEBUG: {1 >>> <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - <core> >>> [core/parser/parse_uri.c:1254]: parse_uri(): uri too short: <<BC><EA><8F>> >>> (3) >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: DEBUG: {1 >>> <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - <core> >>> [core/parser/parse_uri.c:1328]: parse_sip_msg_uri(): bad uri <<BC><EA><8F>> >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: WARNING: >>> {1 <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity >>> [sanity.c:282]: check_ruri_scheme(): failed to parse request uri >>> [<BC><EA><8F>] >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: DEBUG: {1 >>> <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity >>> [sanity_mod.c:254]: w_sanity_check(): sanity checks result: 0 >>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[909]: ERROR: {1 >>> <null> 172.30.154.189 102 ACK >>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - <script>: Malformed >>> SIP request from 172.30.154.189:5060 >>> >>> Still, not sure what the problem is though... >>> >>> BR, >>> George >>> >>> On Wed, 8 Jul 2020 at 09:30, Daniel-Constantin Mierla <miconda(a)gmail.com> >>> wrote: >>> >>>> Hello, >>>> >>>> when the ruri scheme check fails, there should be another debug message >>>> saying that. Have you pasted all log messages for the failure case? >>>> >>>> Cheers, >>>> Daniel >>>> On 07.07.20 22:23, George Diamantopoulos wrote: >>>> >>>> Sorry, I realised I copy pasted wrong log messages for Call 1. Here's >>>> the relevant part showing success for call 1 in contrast with Call 2: >>>> >>>> grep 2a859fcc4e1c8f840191a81d7c16e76d kamailio.log | egrep >>>> 'check_ruri_scheme|w_sanity_check' | grep ACK >>>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[907]: DEBUG: >>>> {1 <null> 172.30.154.189 102 ACK >>>> 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity >>>> [sanity.c:277]: check_ruri_scheme(): check_ruri_scheme entered >>>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[907]: DEBUG: >>>> {1 <null> 172.30.154.189 102 ACK >>>> 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity >>>> [sanity.c:297]: check_ruri_scheme(): check_ruri_scheme passed >>>> Jul 7 18:42:11 lbpub0-stage-lhe0-cn1 /usr/sbin/kamailio[907]: DEBUG: >>>> {1 <null> 172.30.154.189 102 ACK >>>> 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity >>>> [sanity_mod.c:254]: w_sanity_check(): sanity checks result: 1 >>>> >>>> On Tue, 7 Jul 2020 at 21:34, George Diamantopoulos < >>>> georgediam(a)gmail.com> wrote: >>>> >>>>> Hello all, >>>>> >>>>> I'm not 100% sure this is the only culprit in an issue I'm >>>>> investigating, but superficially it appears that RURI scheme sanity module >>>>> checks from the default config (flags 17895 in REQINIT) fail if the >>>>> RURI in an ACK following a 487 includes parameters. Example from two calls >>>>> from a kamailio instance acting as registrar/usrloc server, INVITE RURIs >>>>> are after usrloc lookup: >>>>> >>>>> Call 1: INVITE sip:voip-test-gd@172.17.173.14:5063 SIP/2.0 >>>>> Call 2: INVITE sip:voip-test-user-02@10.2.24.142:32768;line=moo62e08 >>>>> SIP/2.0 >>>>> >>>>> These INVITEs produce no complaints. Later, the same registrar >>>>> produces ACKs to acknowledge 487 (thus, same transaction ACKs) responses >>>>> from the next proxy in the path following a CANCEL: >>>>> >>>>> Call 1: ACK sip:voip-test-gd@172.17.173.14:5063 SIP/2.0 >>>>> Call 2: ACK sip:voip-test-user-02@10.2.24.142:32768;line=moo62e08 >>>>> SIP/2.0 >>>>> >>>>> The next proxy (which produced/relayed the 487) processes the ACK for >>>>> Call 1 successfully, but sanity_check at the proxy drops the request for >>>>> Call 2 with: >>>>> >>>>> DEBUG: {1 <null> 172.30.154.189 102 ACK >>>>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity >>>>> [sanity.c:277]: check_ruri_scheme(): check_ruri_scheme entered >>>>> DEBUG: {1 <null> 172.30.154.189 102 ACK >>>>> 08679c4228983f9e65f3b47f767b6e07(a)voip.domain.com - sanity >>>>> [sanity_mod.c:254]: w_sanity_check(): sanity checks result: 0 >>>>> >>>>> whereas Call 1 seems OK: >>>>> >>>>> DEBUG: {1 <null> 172.30.154.189 102 ACK >>>>> 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity >>>>> [sanity.c:305]: check_required_headers(): check_required_headers entered >>>>> DEBUG: {1 <null> 172.30.154.189 102 ACK >>>>> 2a859fcc4e1c8f840191a81d7c16e76d(a)voip.domain.com - sanity >>>>> [sanity.c:313]: check_required_headers(): check_required_headers passed >>>>> >>>>> Could this be a bug in sanity module? Is there anything one can do in >>>>> config which could result in illegal ACKs being produced for hop-by-hop >>>>> transactions? schema appears to be sip: in both cases... >>>>> >>>>> Thank you. Best regards, >>>>> George >>>>> >>>> >>>> _______________________________________________ >>>> 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 >>>> Funding: https://www.paypal.me/dcmierla >>>> >>>> -- >>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >>> Funding: https://www.paypal.me/dcmierla >>> >>> [View Less]

2 2

Setting up Kamailio as Teams SBC
by Роман С. 08 Jul '20

08 Jul '20

Hello. I'm trying to set up Kamailio as SBC for Teams using https://skalatan.de/en/blog/kamailio-sbc-teams. Setup is completely default (except things mentioned at article), but I use wildcard certificate for TLS. Well, I can't even pass dispatcher: kamcmd dispatcher.list | egrep "RI|FLAG" URI: sip:sip.pstnhub.microsoft.com ;transport=tls FLAGS: IP PRIORITY: 3 … [View More]

3 6

sanity module checks fail for ACK with parameters in RURI - possible bug?
by George Diamantopoulos 08 Jul '20

08 Jul '20

Hello all, I'm not 100% sure this is the only culprit in an issue I'm investigating, but superficially it appears that RURI scheme sanity module checks from the default config (flags 17895 in REQINIT) fail if the RURI in an ACK following a 487 includes parameters. Example from two calls from a kamailio instance acting as registrar/usrloc server, INVITE RURIs are after usrloc lookup: Call 1: INVITE sip:voip-test-gd@172.17.173.14:5063 SIP/2.0 Call 2: INVITE sip:voip-test-user-02@10.2.24.142:… [View More]

2 5

Some Functions not defined in Documentation of Module ims_registrar_pcscf
by Hamid Hashmi 08 Jul '20

08 Jul '20

I have been using a sample code of PCSCF<https://github.com/kamailio/kamailio/blob/master/misc/examples/ims/pcscf/ro…>, where a function "pcscf_assert_identity" is used but it's not defined in module ims_registrar_pcscf documentation. Can you please explain its functionality. The same way other functions are not defined in the documentation. pcscf_assert_called_identity reginfo_handle_notify pcscf_unregister Regards Hamid R. Hashmi

1 0

Kamailio 5.3.4 - Segfault on http_async_client
by José Seabra 07 Jul '20

07 Jul '20

Hello there, Kamailio is terminating with signal 11 and is generating a coredump file. I opened the core file generated by it and it seems to be related to the module http_async_client. In the following link you can find the output of core dump backtrace full: https://pastebin.com/rapPZy9N Below is the http_async_client config that I have: modparam("http_async_client", "workers", 2) modparam("http_async_client", "connection_timeout", 500) modparam("http_async_client", "hash_size", 2048) … [View More]

1 1

Kamailio Buildroot
by Benjamin Marty 07 Jul '20

07 Jul '20

Hello I would like to ask if someone already tried to build Kamailio for the Embedded Distribution Buildroot. I tried it myself but it looks like Kamailio is not that easy to cross compile. Thanks Benjamin

3 3

siptrace not sending data to hepic
by Andrew Chen 07 Jul '20

07 Jul '20

Hi all, I've been assigned a task to investigate why hepic doesn't generate MOS scores ever since we changed the kamailio interface MTU from 9001 -> 1500. During my investigation, this is what I have found out so far: - Putting MTU back to 9001 removes this condition where no MOS was generated. - I did a tcpdump of port 9063 which is the port used to send data to Hepic capture server and saw it reports packets are fragmented: 3:48:55.895505 IP (tos 0x68, ttl 64, id 38049, offset 0, flags … [View More]

2 2

Keep one registration per device(src_ip+port)
by Zhan Bazarov 07 Jul '20

07 Jul '20

Hello. I'd like to keep one registration per device in usrloc table, for other cases I want to store and maintain one contact per AoR. Does kamailio support it out of the box? For second case save("location", "0x01"); working fine. But how about first case? for example, if one of device located at home, and second located at office. - there are two different network and NAT - how I can make both devices ringing when INVITE is sent? If I do classic multiAor scenario - it can lead to issues … [View More]

1 0

Nathelper ping from dedicated kamailio instance
by Angelo Pellegrinon 07 Jul '20

07 Jul '20

Hi! I am new to kamailio so please forgive me if I write some errors. I am configuring some kamailio instances in different environments to handle big traffic and I have a problem with usrloc + db + nathelper. What have I done: 1. Configured a kamailio instance to handle sip registers with usrloc to database with database mode 2 (memory + db), disabled nathelper pings. This is ok and performs well; 2. Configured a kamailio instance to handle nathelper ping. Same usrloc + db + nathelper … [View More]

1 0

rtpengine with mutiple IPs on the same network.
by David Villasmil 07 Jul '20

07 Jul '20

Hello guys, I have rtpengine running on a box with ips like this: external: 172.10.0.10 (advertise 10.10.10.1) internal: 172.10.0.11 Both Ips are on the same network. When i do rtpengine_manage i set "direction=external direction=internal", and the SDP offer is correct. The problem I'm having is that when rtp tried to send to the internal network, the OS (i believe) selects the first interface (external) ip 172.10.0.10 to reach the internal network. This I suppose is not a great deal, but I … [View More]

3 5

Retrieving cert details from tls peer
by Mark Boyce 07 Jul '20

07 Jul '20

Hi all Been trying to grab the TLS cert details from incoming connections, but failing :-( So with lines just before AUTH is called like this; if (proto == TLS) { xlog("L_INFO", "TLSDUMP $ci peer_subject : $tls_peer_subject\n"); Gets met with a log line line this; INFO: tls [tls_server.c:431]: tls_accept(): tls_accept: new connection from 1.2.3.4:11797 using TLSv1.2 ECDHE-RSA-AES256-GCM-SHA384 256 INFO: tls [tls_server.c:434]: tls_accept(): tls_accept: local socket: … [View More]5.6.7.8:5061 INFO: tls [tls_server.c:445]: tls_accept(): tls_accept: client did not present a certificate ... INFO: tls [tls_select.c:168]: get_cert(): Unable to retrieve peer TLS certificate from SSL structure This is with verify_certificate and require_certificate set to no in tls.cfg If I try and set the following in tls.cfg [server:default] method = TLSv1.2+ verify_certificate = no require_certificate = yes I see in the logs; INFO: tls [tls_domain.c:303]: ksr_tls_fill_missing(): TLSs<default>: tls_method=22 INFO: tls [tls_domain.c:315]: ksr_tls_fill_missing(): TLSs<default>: certificate='/etc/kamailio/tls-certs/cert.pem' INFO: tls [tls_domain.c:322]: ksr_tls_fill_missing(): TLSs<default>: ca_list='(null)' INFO: tls [tls_domain.c:329]: ksr_tls_fill_missing(): TLSs<default>: crl='(null)' INFO: tls [tls_domain.c:333]: ksr_tls_fill_missing(): TLSs<default>: require_certificate=1 INFO: tls [tls_domain.c:340]: ksr_tls_fill_missing(): TLSs<default>: cipher_list='(null)' INFO: tls [tls_domain.c:347]: ksr_tls_fill_missing(): TLSs<default>: private_key='/etc/kamailio/tls-certs/privkey.pem' INFO: tls [tls_domain.c:351]: ksr_tls_fill_missing(): TLSs<default>: verify_certificate=0 INFO: tls [tls_domain.c:354]: ksr_tls_fill_missing(): TLSs<default>: verify_depth=9 NOTICE: tls [tls_domain.c:1095]: ksr_tls_fix_domain(): registered server_name callback handler for socket [:0], server_name='<default>' ... INFO: tls [tls_domain.c:692]: set_verification(): TLSs<default>: Client MUST present valid certificate INFO: tls [tls_domain.c:303]: ksr_tls_fill_missing(): TLSc<default>: tls_method=20 INFO: tls [tls_domain.c:315]: ksr_tls_fill_missing(): TLSc<default>: certificate='(null)' INFO: tls [tls_domain.c:322]: ksr_tls_fill_missing(): TLSc<default>: ca_list='(null)' INFO: tls [tls_domain.c:329]: ksr_tls_fill_missing(): TLSc<default>: crl='(null)' INFO: tls [tls_domain.c:333]: ksr_tls_fill_missing(): TLSc<default>: require_certificate=1 INFO: tls [tls_domain.c:340]: ksr_tls_fill_missing(): TLSc<default>: cipher_list='(null)' INFO: tls [tls_domain.c:347]: ksr_tls_fill_missing(): TLSc<default>: private_key='(null)' INFO: tls [tls_domain.c:351]: ksr_tls_fill_missing(): TLSc<default>: verify_certificate=1 INFO: tls [tls_domain.c:354]: ksr_tls_fill_missing(): TLSc<default>: verify_depth=9 INFO: tls [tls_domain.c:692]: set_verification(): TLSc<default>: Server MUST present valid certificate ... ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417C086:SSL routines:tls_process_client_certificate:certificate verify failed Which looks like verification is being enabled when I add require? Would someone be kind enough to point out what I am missing please? (Assuming it’s not a bug :-) Thanks Mark -- Mark Boyce Dark Origins Ltd [View Less]

2 8

About routing calls
by ahmed moghazy 07 Jul '20

07 Jul '20

Excuse me i already made integration for kamailio with asterisk but i want to make sip calls through kamailio and media fuctions through asterisk , i want kamailio to make calls away from asterisk , i just wanted asterisk to use its dahdi module .’ Thank you.

2 1

Kamailio Push Notification for iOS13
by william anderson 07 Jul '20

07 Jul '20

Hello, Yesterday, there was a new release of Linphone IOS app, but i need help from some kamailio expert how to add PUSH notification to the application to receive incoming calls when the phone is not in use (wake up required) from a kamailio server. I have read on internet old tutorials (tsilo module) but most of them are incomplete or do not explain in an easy way how to do it (for an average kamailio user). Anyone could do an easy to understand tutorial ? Regards, William Anderson

2 1

Looking for a VoIP & Kamailio Dev-Ops
by Henning Westerholt 06 Jul '20

06 Jul '20

Dev-Ops VoIP & Kamailio GILAWA is looking for a new colleague to join our team! Tasks: Deploying, developing, and integrating our software based on Kamailio Testing and validation of our customer solutions Solve complex customer problems in the VoIP field Fix Bugs and create extensions in the Kamailio project Requirements: Good background in VoIP telephony and Kamailio Stack Experience with signalling standards (SIP, RTP etc..) and troubleshooting Knowledge and experience in one of the … [View More]

3 3

Kamailio 5.3.5, usrloc: Error inserting location (integer out of range)
by Stefan 03 Jul '20

03 Jul '20

Hi Maillist Kamailio-Version: 5.3.5 built from source with postgres-db. I get an Error if I try to add a static location with kamctl, see my example: kamctl ul add 1234 sip:1234@10.10.10.10:5060 NULL NULL Log output: Jul 1 09:28:20 kamailio-srv02 /sbin/kamailio[3892]: ERROR: db_postgres [km_dbase.c:266]: db_postgres_submit_query(): 0x7fbe5398c148 PQsendQuery Error: ERROR: integer out of range#012 Query: insert into location (username,contact,expires,q,callid,cseq,flags,cflags,… [View More]user_agent,path,methods,last_modified,ruid,reg_id,server_id,connection_id,keepalive,partition ) values ('1234','sip:1234@10.10.10.10:5060','1970-01-01 00:00:00',1.000000 ,'dfjrewr12386fd6-343(a)kamailio.rpc',1,0,0,'kamailio SIP Router - RPC Server','NULL',2147483648,'1970-01-01 00:00:00','ulcx-5ef99724-f37-1',0,0,0,0,22) truncated output of kamctl ul show: "AoRs": [{ "Info": { "AoR": "1234", "HashID": 924651761, "Contacts": [{ "Contact": { "Address": "sip:1234@10.10.10.10:5060", "Expires": "permanent", "Q": 1, "Call-ID": "dfjrewr12386fd6-343(a)kamailio.rpc", "CSeq": 1, "User-Agent": "kamailio SIP Router - RPC Server", "Received": "[not set]", "Path": "NULL", "State": "CS_NEW", "Flags": 0, "CFlags": 0, "Socket": "[not set]", "Methods": -2147483648, "Ruid": "ulcx-5ef99724-f37-1", "Instance": "[not set]", "Reg-Id": 0, "Server-Id": 0, "Tcpconn-Id": 0, "Keepalive": 0, "Last-Keepalive": 0, "Last-Modified": 0 } }] } }, { So the log tells me: integer out of range. I think it's this one: "Methods": -2147483648, This let's me thinking that this is a bug. Someone else? Regards Stefan [View Less]

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403 Forbidden error You must register first with a S-CSCF
by Pavithra Mohanraja 03 Jul '20

03 Jul '20

Hi, I am facing below error when i try to make a call using pjsip .. Registration is successful. I am following the open 5gs VolTE setup. I have not changed any configurations in any of the config files. But I am facing below issue .. Can you please help me what could be the reason. *reason=403 (Forbidden - You must register first with a S-CSCF)* Could anyone please help me .. I am ready to provide anyother info if required . Thanks, Pavithra

1 0

Error in SCSCF
by Pavithra Mohanraja 02 Jul '20

02 Jul '20

Hi , I am facing below error in s-cscf when i send a BYE from alice to bob in kamailio IMS. The problem is in only call termination .Kindly help . Error log in scscf : Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/mem/q_malloc.c:526]: qm_free(): freeing frag. 0x7f4746f89148 alloc'ed from core: core/rvalue.c: rval_new_empty(224) Jun 30 08:40:22 scscf kamailio[557]: DEBUG: <core> [core/socket_info.c:630]: grep_sock_info(): checking if host==us: 11==11 && [10.244.… [View More]

2 4

Sql_result_free() not cleaning properly
by Duarte Rocha 01 Jul '20

01 Jul '20

Greetings, I'm trying to clean a SQL result using "sql_result_free()" but i'm not having success. Can you please verify if i'm doing anything wrong here? This is my code : xerr("Pre clean [0] = Iterator 0 = $xavp(Cli_PICKER_Default_Search_Result[0]=>Code)"); sql_result_free("Cli_PICKER_Default_Search_Result"); xerr("Pos clean [0] = Iterator 0 = $xavp(Cli_PICKER_Default_Search_Result[0]=>Code)"); The result is the following : ERROR: … [View More]

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Re: [SR-Users] [EXT] RE: kamailio 522 S-CSCF failing to start -UPDATE
by Martin W Woscek 30 Jun '20

30 Jun '20

Problem: The mysql db issues with the scscf db include: table version, disappearing impu and/or UE data retained in tables when none are registered Workarround: In order to get over those various mysql issues listed causing the S-CSCF not to come up properly as boot up, 1. create a fresh new scscf database (kamadbctl [drop|create] scscf and import the msql script from the scscf configs) 2. back up the new scscf db 3. modify the init.d startup script for the scscf to restore the fresh … [View More]db upon each start/restart of the S-CSCF /etc/init.d/kamailio_scscf .. case "$1" in start|debug) mysql -u root --database=scscf < /etc/kamailio/ims_databases/scscf.sql The S-CSCF has been stable and dependable on each boot. _Martin From: Henning Westerholt <hw(a)skalatan.de> Sent: Monday, June 8, 2020 4:50 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Cc: Martin W Woscek <mwoscek(a)mitre.org> Subject: [EXT] RE: kamailio 522 S-CSCF failing to start Hello, the error message below is from the DB startup version check. So do you have sometimes DB errors on this virtual machine? Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/> From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of Martin W Woscek Sent: Monday, June 8, 2020 2:46 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> Subject: [SR-Users] kamailio 522 S-CSCF failing to start Hello, There are times that our S-CSCF module just wont start. I’ve tried correlating the issue with some operation such as not de-registering an IMS client (Boghe or imsdroid) prior to shutting down the Kamailio server. But this isn’t always the case. Being that we are running this an a Ubuntu Virtual Machine, simply overwriting the image with a backed up VM is the temporary solution. I’ve dropped and created the db for S (kamdbctl) and rearranged the order of loading the presence module with same result. Status of the S-CSCF daemon on failure to start: # systemctl status kamailio_scscf ● kamailio_scscf.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio_scscf; generated) Active: inactive (dead) Docs: man:systemd-sysv-generator(8) Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jun 08 08:22:55 ubuntu kamailio_scscf[16495]: * already running Jun 08 08:22:55 ubuntu kamailio_scscf[16495]: ...done. Jun 08 08:22:55 ubuntu systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Jun 08 08:24:03 ubuntu systemd[1]: Stopping LSB: Start the Kamailio SIP proxy server... Jun 08 08:24:03 ubuntu kamailio_scscf[16517]: * Stopping Kamailio IMS S-CSCF: kamailio_scscf Jun 08 08:24:03 ubuntu kamailio_scscf[16517]: ...done. Jun 08 08:24:03 ubuntu systemd[1]: Stopped LSB: Start the Kamailio SIP proxy server. root@ubuntu:~/IMS_manual_scripts# systemctl start kamailio_scscf root@ubuntu:~/IMS_manual_scripts# systemctl status kamailio_scscf ● kamailio_scscf.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio_scscf; generated) Active: active (exited) since Mon 2020-06-08 08:27:23 EDT; 58s ago Docs: man:systemd-sysv-generator(8) Process: 16543 ExecStart=/etc/init.d/kamailio_scscf start (code=exited, status=0/SUCCESS) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: <core> [db.c:457]: db_check_table_version(): invalid version 0 for table presentity found, expected 5 (check table structure and ta Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: presence [presence.c:367]: mod_init(): error during table version check Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: <core> [core/sr_module.c:898]: init_mod(): Error while initializing module presence (/usr/local/lib64/kamailio/modules/presence.so) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: WARNING: <core> [core/mem/q_malloc.c:480]: qm_free(): WARNING: free(0) called from cdp_avp: cdp_avp_mod.c: cdp_avp_destroy(226) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jun 08 08:27:23 ubuntu kamailio_scscf[16543]: * already running Jun 08 08:27:23 ubuntu kamailio_scscf[16543]: ...done. Jun 08 08:27:23 ubuntu systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Any insight is appreciated. Thanks, _Martin [View Less]

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default config and options for Kamailio 5.4.x
by Daniel-Constantin Mierla 30 Jun '20

30 Jun '20

Hello, with the release of Kamailio 5.4.0 approaching, I opened an item in the tracker to collect suggestions about what should be tuned in the default config file. If you feel like something should be there, add your comment to: * https://github.com/kamailio/kamailio/issues/2381 Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

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Re: [SR-Users] RTP voice failover
by Yufei Tao 30 Jun '20

30 Jun '20

Hi, RTP FO is possible, e.g. with keepalived+rtpengine+Redis as people suggested. Here's the guide for configuring it which I find very useful: https://github.com/sipwise/rtpengine/wiki/Redis-keyspace-notifications A possible way is to have a pair of rtpengine instances in master-slave setup. When master goes down, RTP continues on slave (new master) after a few seconds gap, when the virtual IP(s) migrate over. Another useful link: https://www.kamailio.org/events/2016-KamailioWorld/Day2/20-… [View More]

1 0

kamailio and rtpengine in docker
by David Villasmil 29 Jun '20

29 Jun '20

Hello guys, I've got a kamailio on a docker and kamailio on another docker on different boxes. The issue i'm having is that when kamailio tries to enable rtpengine it tries pinging it. The ping arrives at the actual container where rtpengine is running, but for some reason i haven't figured out, rtpengine (or the container) is not responding to the ping. I'm writing here in the hopes that someone has already implemented this an come across this and a solved (for sure) I'd appreciate some … [View More]

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Valid URL characters in the RURI
by Sam Ware 29 Jun '20

29 Jun '20

I'm using Sanity Checks. Is an underscore (_) not allowed in the host portion of the RURI? I'm on version 5.3. I reviewed the pase_uri function in the core parse_uri.c but I am not super proficient in C. case URI_HOST_P: switch(*p) { check_host_end; default: if(!isalnum(*p) && (*p != '.') && (*p != '-') && !uri_host_char_allowed(*p)) { goto error_bad_host; } } break; I see it calls out periods (.) and dashes … [View More]

2 1

Changing Debug Level on the fly
by Sam Ware 29 Jun '20

29 Jun '20

I am trying to change the debug level without restarting the service. My research says this is right but I get an error message. $ sudo -u kamailio kamcmd dbg.set_mod_level core 2 error: 500 - cannot store parameter -- Sam D Ware

2 1

SIP OPTIONS between UE and P-CSCF
by BALL SUN 29 Jun '20

29 Jun '20

Hi We have setup UE with Kamailio IMS, and after the registration is successful. The 2nd SIP REGISTER is sent from UE to P-CSCF through IPSEC tunnel. But we found that the SIP OPTIONS (I believed it is a health check) from P-CSCF to UE is not through the IPSEC tunnel. our question is 1. Should SIP OPTIONS run inside the IPSEC tunnel or it should be outside? 2. if it can be configured to run inside the IPSEC tunnel, what configuration can we change to achieve it? - RBK

2 3

How to add parameter ;user=phone in RURI
by Minh Phan 28 Jun '20

28 Jun '20

Hi List, I'm having trouble adding the parameter user=phone in the RURI. I have tried to use the function add_uri_param from siputils module but it doesn't work (see below). Here is the code I use to add user=phone parameter to FROM, TO and RURI uac_replace_from("", "sip:$var(normalized_number)@sip.domain.com ;user=phone"); uac_replace_to("", "sip:$var(normalized_number)@sip.domain.com;user=phone"); $rU = $var(normalized_number); *add_uri_param*("user=phone"); Thanks in advance for any … [View More]

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RTP voice failover
by Evgeniy 27 Jun '20

27 Jun '20

Is that possible to implement RTP voice fail-over? In case RTP node goes down - the call should not be interrupted. This is kinda strange - but client want to have super redundant system. For example use 2 RTP nodes and simultaneously send 2 RTP stream over 2 nodes (yes - network traffic will increase) Or maybe is a way to recover UDP connections after a crash and replace node state. Maybe use a Kubernetes tools for that purpose ? Here what i found in mailing-list - https://lists.… [View More]

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Kamailio IMS with ENUM
by BALL SUN 26 Jun '20

26 Jun '20

Hi Is there any information that can help to setup ENUM query in Kamailio IMS? what we wanted to achieve is to when A in domain A (the local domain) makes an INVITE to B, it will check the domain of B from ENUM backend. if it is not on the local domain, it will route to remote IMS core. Thanks - RBK

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Does kamailio sip domain makes sense if i defined some alias
by PICCORO McKAY Lenz 26 Jun '20

26 Jun '20

in kamctlrc i defined an DOMAIN but later that ip was updated.. so my question: it make sense inclusivelly if already defined more allias in kamailiorc.cfg file? in case of more complicated response,, anybody can explain me about that why? -- Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

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Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag) -RESOLVED
by Martin W Woscek 26 Jun '20

26 Jun '20

Hello, Just wanted to update what fixed our issue. The "contact_user" which is "asterisk" that we were using for the Kamailio context for outgoing SIP trunk/channel in the pjsip.conf file of the Asterisk 16 server was not a valid user in the IMS HSS database (FHoSS implementation). However not being an expert in the IMS, IMHO, the IMS should have kicked out this INVITE as soon as credentials were checked at the front door of the IMS, but rather Kamailio sent it from I-CSCF to S-CSCF to P-CSCF,… [View More] which is the correct flow, until the P-CSCF sent it back to I-CSCF which then resulted in a 604 HSS user unknown being sent thru the reverse direction back to Asterisk. Anyway, its always good to end Friday on a high note. Thanks, _Martin From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Martin W Woscek Sent: Wednesday, June 24, 2020 1:32 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>; miconda(a)gmail.com Subject: Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag) Just to clarify what we are seeing, here is a packet from the INVITE from the S-CSCF (6060) to the P-CSCF(4060). I have eliminated the S-CSCF as this packet looks correct. The P-CSCF is not handling the request URI correctly, and simply sends it back to the I-CSCF where I-CSCF cannot find the user in the HSS based on the ruri updated by the S-CSCF with the local IP address of the UE (192.168.20.1). The P-CSCF should send this INVITE to the UE (703222888(a)192.168.20.1<mailto:703222888@192.168.20.1>) No. Arrival Time Source Destination Source Port Destination Port SIP to tag SIP from tag Method Record-Route Userinfo Request-URI Media Port Protocol Length Request-URI Host Port Owner Username Media Type Media Format Origin-Host Type slice_type SIP from address User Part SIP from address Host Part SIP to address User Part SIP to address Host Part Contact URI User Part Contact URI Host Part Request-URI User Part Request-URI Host Part Request-URI Host Port Info 1553 Jun 17, 2020 09:31:21.722451703 Eastern Daylight Time 192.168.20.128 192.168.20.128 6060 4060 d490a30c-42a8-4fc0-9fe9-b7a21537386d INVITE mt sip:7032228888@192.168.20.1:63871;transport=udp 10626,13370 SIP/SDP 2263 63871 - audio,video ITU-T G.711 PCMU,DynamicRTP-Type-101,0,101,101,DynamicRTP-Type-99,99,99 7033331111 asterisk.irisims.org 7032228888 ims522.irisims.org asterisk 192.168.20.131 7032228888 192.168.20.1 63871 Request: INVITE sip:7032228888@192.168.20.1:63871;transport=udp | Frame 1553: 2263 bytes on wire (18104 bits), 2263 bytes captured (18104 bits) on interface any, id 0 Linux cooked capture Internet Protocol Version 4, Src: 192.168.20.128, Dst: 192.168.20.128 User Datagram Protocol, Src Port: 6060, Dst Port: 4060 Session Initiation Protocol (INVITE) Request-Line: INVITE sip:7032228888@192.168.20.1:63871;transport=udp SIP/2.0 Method: INVITE Request-URI: sip:7032228888@192.168.20.1:63871;transport=udp Request-URI User Part: 7032228888 Request-URI Host Part: 192.168.20.1 Request-URI Host Port: 63871 [Resent Packet: False] Message Header Record-Route: <sip:mt@192.168.20.128:6060;lr=on;ftag=d490a30c-42a8-4fc0-9fe9-b7a21537386d;did=7d2.e0f1> Route: <sip:term@pcscf522.irisims.org;lr> Via: SIP/2.0/UDP 192.168.20.128:6060;branch=z9hG4bK285b.2a57abaad57cfd55bc18a23d4e9cefc8.0 Via: SIP/2.0/UDP 192.168.20.128;branch=z9hG4bK285b.793ba7dcf9fcd18ddef225b83896340e.1 Via: SIP/2.0/UDP 192.168.20.131:5060;received=192.168.20.131;rport=5060;branch=z9hG4bKPjef451437-0a4f-4732-ac54-318740956335 From: <sip:7033331111@asterisk.irisims.org>;tag=d490a30c-42a8-4fc0-9fe9-b7a21537386d To: <sip:7032228888@ims522.irisims.org> Contact: <sip:asterisk@192.168.20.131:5060> Call-ID: 39c15c42-b0b7-48b5-b4bf-ac7a99fbe660 [Generated Call-ID: 39c15c42-b0b7-48b5-b4bf-ac7a99fbe660] CSeq: 19314 INVITE Route: <sip:ims522.irisims.org:5060;lr> Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 Max-Forwards: 68 User-Agent: Asterisk PBX 16.6.0 Content-Type: application/sdp Content-Length: 1121 Message Body From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of Martin W Woscek Sent: Wednesday, June 24, 2020 11:21 AM To: miconda(a)gmail.com<mailto:miconda@gmail.com>; Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> Subject: Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag) Hi, The issue is the P should be sending to the UE not back to the I like the failed flow: Asterisk16-[phone] --->ICSCF--->S-CSCF--->P-CSCF--->I-CSCF ---> [604 HSS user unknown] to Asterisk16[phone] The successful call should look like: Asterisk16-[phone] --->ICSCF--->S-CSCF--->P-CSCF--->Kamailio-[UE] Does anyone have a example P-cscf scripts including the mt.cfg or other script file that make this work? The reverse calls work just fine. Thanks, _Martin From: Daniel-Constantin Mierla <miconda(a)gmail.com<mailto:miconda@gmail.com>> Sent: Wednesday, June 24, 2020 4:53 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>; Martin W Woscek <mwoscek(a)mitre.org<mailto:mwoscek@mitre.org>> Subject: [EXT] Re: [SR-Users] Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag) Hello, Kamailio is not generating the 200ok for INVITE (calls), it just sends out what it received from end point, after consuming own Via header (of course, other headers can be changed based on config rules, but Kamailio is not in charge of setting To-tag). Cheers, Daniel On 19.06.20 21:19, Martin W Woscek wrote: Hello all, SIP/UE (boghe or imsdroid) client registered to Kamailio makes call to an Asterisk registered SIP phone is successful: [UE]-Kamailio--->INVITE--->Asterisk16-[phone] [UE]-Kamailio--->200 OK with to_tag--->Asterisk16-[phone] But in reverse direction for a call, Kamailio does not return the SIP OK so no to_tag is sent so call fails to ring and complete: [phone]-Asterisk16 --->INVITE--->Kamailio Where the UE device never receives the INVITE, Asterisk never gets and 200 OK message with the to_tag from the I-CSCF, the call flow itself gets lost in Kamailio, where the P-CSCF sends final INVITE to I-CSCF and and ultimately a 604 HSS user unknown message is sent back to Asterisk from the I-CSCF. Basically Im using the default "sample" configs for both the P and the I-CSCF. Our sauce is in the S-CSCF for out going calls that originate by a registered UE. Any insight or sample Kamailio configuration that Im lacking? Has anyone done this and could share the asterisk and Kamailio script snippets that make it possible. 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 -- Daniel-Constantin Mierla -- www.asipto.com<http://www.asipto.com> www.twitter.com/miconda<http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda> Funding: https://www.paypal.me/dcmierla [View Less]

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Question about textops
by Mojtaba 26 Jun '20

26 Jun '20

Hello, Some confusing things while developing in Kamailio: In is_method_f function in textops module, the code checks the HDR_CSEQ_F in msg also, and if the method name appears in CSEQ_F, The result is returned TRUE. These question will arise why in this function, the CSEQ field in message will check? if(parse_headers(msg, HDR_CSEQ_F, 0)!=0 || msg->cseq==NULL) { LM_ERR("cannot parse cseq header\n"); return -1; /* should it be 0 ?!?! */ } if(m->s==0) return (get_cseq(msg)->method_id&… [View More]

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rtpengine inserted random line into sdp
by Andrew Chen 26 Jun '20

26 Jun '20

Hi forum, I'm starting my rtpengine project and I'm facing a strange problem with rtpengine. I am seeing this in the SDP part of the INVITE: a=rtcp:52021 a=rtcp-mux 2001:470:7:3A7:0:0:0:2 a=direction:active a=oldmediaip:54.153.25.234 As you can see there is a random insert of the local kamailio IP in there and I'm have tried the following to remove it but failed to succeed: - Remove in the offer the combination of ICE and replace-session-connection flags - Switched between … [View More]

2 4

Advertise concept
by Social Boh 25 Jun '20

25 Jun '20

Hello, maybe i don't understood completely the use of Advertise parameter. We use it only if Kamailio is behind a NAT and Kamailio have to comunicate over is PublicIP with provider, other SIP proxy and PBX but never to accept Users REGISTER. Is it right? Regards -- --- I'm SoCIaL, MayBe

2 3

Server error on LIR select next S-CSCF
by BALL SUN 25 Jun '20

25 Jun '20

Hi I have a setup using Kamailio IMS, and when trying to make a SIP call between clients, S-CSCF returns " SIP/2.0 500 Server error on LIR select next S-CSCF", do you have any idea on how to fix? - RBK

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kamailio - 5.3.2 - Possible memleak in Core Module
by José Seabra 25 Jun '20

25 Jun '20

Hello there. During my stress tests against our kamailio servers I detected that it was running out of shm memory. edge-sip-proxy[6119]: ERROR: ESP_LOG: 10405249-1580(a)10.225.121.206: <core> [core/mem/q_malloc.c:298]: qm_find_free(): qm_find_free(0x7ff370920000, 4224); Free fragment not found! edge-sip-proxy[6119]: ERROR: ESP_LOG: 10405249-1580(a)10.225.121.206: <core> [core/mem/q_malloc.c:432]: qm_malloc(): qm_malloc(0x7ff370920000, 4224) called from core: core/sip_msg_clone.c: … [View More]sip_msg_shm_clone(496), module: core; Free fragment not found! edge-sip-proxy[6119]: ERROR: ESP_LOG: 10405249-1580(a)10.225.121.206: <core> [core/sip_msg_clone.c:499]: sip_msg_shm_clone(): could not allocate shared memory from shm pool edge-sip-proxy[6119]: ERROR: ESP_LOG: 10405249-1580(a)10.225.121.206: tm [t_lookup.c:1293]: new_t(): out of mem: edge-sip-proxy[6119]: ERROR: ESP_LOG: 10405249-1580(a)10.225.121.206: tm [t_lookup.c:1439]: t_newtran(): new_t failed edge-sip-proxy[6119]: ERROR: ESP_LOG: 10405249-1580(a)10.225.121.206: sl [sl_funcs.c:392]: sl_reply_error(): stateless error reply used: I'm terribly sorry, server error occurred (1/SL) The output of kamctl stats shmem confirms that kamailio is running out of memory: { "jsonrpc": "2.0", "result": [ "shmem:fragments = 36427", "shmem:free_size = 14123552", "shmem:max_used_size = 268435456", "shmem:real_used_size = 254311904", "shmem:total_size = 268435456", "shmem:used_size = 215787904" ], "id": 10901 } Then I was check what was the module that were consuming more memory with kamcmd mod.stats all shm and the output of this command showed me that Core module was consuming most of shm memory, as you can see bellow is the output of this command shows that Core - build_req_buf_from_sip_req is consuming most of shm. Module: core { sip_msg_shm_clone(496): 4872856 create_avp(175): 24768 msg_lump_cloner(986): 154376 xavp_new_value(106): 780896 build_req_buf_from_sip_req(2187): 183984624 counters_prefork_init(211): 36864 cfg_clone_str(130): 96 cfg_shmize(217): 848 main_loop(1303): 8 init_pt(106): 8 init_pt(105): 8 init_pt(104): 5256 register_timer(995): 232 cfg_register_ctx(47): 64 init_tcp(5021): 8192 init_tcp(5015): 32768 init_tcp(5007): 8 init_tcp(5000): 8 init_tcp(4993): 8 init_tcp(4987): 8 init_tcp(4975): 8 init_avps(90): 8 init_avps(89): 8 init_dst_blacklist(438): 16384 init_dst_blacklist(430): 8 timer_alloc(498): 96 init_dns_cache(361): 8 init_dns_cache(352): 16384 init_dns_cache(344): 16 init_dns_cache(336): 8 init_timer(267): 8 init_timer(266): 16384 init_timer(265): 8 init_timer(264): 8 init_timer(253): 8 init_timer(221): 8 init_timer(210): 278544 init_timer(209): 8 init_timer(197): 8 cfg_child_cb_new(829): 64 sr_cfg_init(361): 8 sr_cfg_init(354): 8 sr_cfg_init(347): 8 sr_cfg_init(335): 8 sr_cfg_init(323): 8 qm_shm_lock_init(1202): 8 Total: 190229920 } Then I stopped the sipp that was running against kamailio in order to see if the memory consumed by *build_req_buf_from_sip_req* decreased but it didn't, so, it seems that there is a memleak in this version. If the information that I'm sending in this email is not enough to identify the root cause of this, please let me know if there is anything else that I can do here to help identify better what is the main reason. Than you Best Regards -- Cumprimentos José Seabra [View Less]

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siptrace: methods blacklist modparam
by Nuno Ferreira 25 Jun '20

25 Jun '20

Hello, We are a heavy Homer/Hepic (tks QXIP team) users and on our setup, there are some SIP methods we don’t want to get into the database or send to the capture server because there’s no value-added for call troubleshooting purposes (OPTIONS, KDMQ, HTTP (from JSONRPC)…) Typically we achieve the message filtering using siptrace in “manual” mode and control what to send from the script. This works fine for some situations, but there are a few we need to have the automatic mode on: - … [View More]

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Re: [SR-Users] Kamailio-5.3.3. NAT and RTP not working
by supreeth herle 25 Jun '20

25 Jun '20

Please try to keep the mailing list in loop so that it helps others as well facing this issue. I believe this was resolved in 5.3.3 branch not sure why you are facing this issue. Anyways, make sure that modules loaded in kamailio.cfg of PCSCF is in following order loadmodule "ims_dialog" loadmodule "ims_usrloc_pcscf" #!ifdef WITH_IPSEC loadmodule "ims_ipsec_pcscf" #!endif loadmodule "ims_registrar_pcscf" Best Regards, Supreeth On Wed, 24 Jun 2020 at 13:36, Pavithra Mohanraja <… [View More]pavimohan3004(a)gmail.com> wrote: > Hi Supreeth, > > I am getting the below error > > ms_usrloc_pcscf > [/usr/lib/x86_64-linux-gnu/kamailio/modules/ims_usrloc_pcscf.so] > 0(455) DEBUG: ims_registrar_pcscf [ims_registrar_pcscf_mod.c:264]: > mod_init(): Successfully bound to PCSCF Usrloc module > 0(455) DEBUG: <core> [core/sr_module.c:632]: find_mod_export_record(): > found export of <bind_ims_ipsec_pcscf> in module ims_ipsec_pcscf > [/usr/lib/x86_64-linux-gnu/kamailio/modules/ims_ipsec_pcscf.so] > 0(455) ERROR: ims_ipsec_pcscf [cmd.c:85]: bind_ipsec_pcscf(): > configuration error - trying to bind to ipsec pscscf module before being > initialized > 0(455) ERROR: <core> [core/sr_module.c:849]: init_mod(): Error while > initializing module ims_registrar_pcscf > (/usr/lib/x86_64-linux-gnu/kamailio/modules/ims_registrar_pcscf.so) > ERROR: error while initializing modules > 0(455) ERROR: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:305]: mod_destroy(): > Error destroying spi generator > 0(455) ERROR: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:309]: mod_destroy(): > Error destroying port generator > 0(455) DEBUG: tm [t_funcs.c:84]: tm_shutdown(): start > 0(455) DEBUG: tm [t_funcs.c:87]: tm_shutdown(): emptying hash table > 0(455) DEBUG: tm [t_funcs.c:89]: tm_shutdown(): removing semaphores > 0(455) DEBUG: tm [t_funcs.c:91]: tm_shutdown(): destroying tmcb lists > > > Kindly help > > On Wed, Jun 24, 2020 at 5:03 PM supreeth herle <herlesupreeth(a)gmail.com> > wrote: > >> There is no other IPSec configuration needed apart from running Kamailio >> P-CSCF and UE sending a SIP Register with Security-Client information for >> IPSec creation. Kamailio IMS IPSec module creates IPSec tunnels using >> Netlink library. >> >> Best Regards, >> Supreeth >> >> On Wed, 24 Jun 2020 at 13:29, Pavithra Mohanraja <pavimohan3004(a)gmail.com> >> wrote: >> >>> Hi , >>> >>> Could you please tell how to configure ipsec (general configuration not >>> related to kamailio). >>> >>> Thanks, >>> Pavithra >>> >>> On Wed, Jun 24, 2020 at 2:12 PM supreeth herle <herlesupreeth(a)gmail.com> >>> wrote: >>> >>>> reason=404 (Not Found - destination user not found on this S-CSCF)] >>>>> >>>> >>>> Please check your IMPU for the IMS user in the IMS HSS (e.g. FHoSS or >>>> any other HSS you are using to enter IMS user details). SCSCF is not able >>>> to find a registered user with that IMPU (i am assuming that you are >>>> getting this response while calling) >>>> >>>> Increase the debug level on S-CSCF and check out which IMPU is being >>>> referred by SCSCF >>>> >>>> Best Regards, >>>> Supreeth >>>> >>>> On Wed, 24 Jun 2020 at 10:26, Pavithra Mohanraja < >>>> pavimohan3004(a)gmail.com> wrote: >>>> >>>>> Hi, >>>>> >>>>> I am facing below error in kamailio-ims 5.3.4 version. >>>>> >>>>> reason=404 (Not Found - destination user not found on this S-CSCF)] >>>>> >>>>> could you please help me how should i proceed with and where the >>>>> registration details will be stored in kamailio ims. >>>>> >>>>> It fails at this line in kamailio.cfg file in scscf . >>>>> >>>>> route[FINAL_TERM] { >>>>> if (lookup("location")) { >>>>> if (uri==myself) { >>>>> if (!t_newtran()) { >>>>> sl_reply_error(); >>>>> exit; >>>>> } >>>>> xlog("Destination Not Found ----\n"); >>>>> t_reply("404","Not Found - destination user >>>>> not found on this S-CSCF"); >>>>> exit; >>>>> } >>>>> >>>>> here lookup(location) means what it means. Could you please help me >>>>> >>>>> >>>>> >>>>> Thanks, >>>>> Pavithra >>>>> >>>>> On Thu, May 14, 2020 at 9:59 PM supreeth herle < >>>>> herlesupreeth(a)gmail.com> wrote: >>>>> >>>>>> Hello Pavithra, >>>>>> >>>>>> I have written a guide for how to integrate an IMS using Kamailio >>>>>> with open5gs EPC, here is the link >>>>>> https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/. You can >>>>>> find steps to setup rtpengine in that guide. >>>>>> >>>>>> Regarding requirement of rtpengine, its mandatory if you want to use >>>>>> the scripts in that example folder without modification. Also, read the >>>>>> READE of sipwise/rtpengine github repo, its very informative (in a gist >>>>>> rtpengine offer much more feature than rtpproxy) >>>>>> >>>>>> why can't we have sip client to sip client rtps flowing like how it >>>>>>> is happening without NAT. >>>>>>> >>>>>> >>>>>> For this, i would definitely suggest to read about SIP working. This >>>>>> is a perfect link https://github.com/onmyway133/blog/issues/284 >>>>>> which offers extensive material (short and very interesting to read with >>>>>> examples) related to SIP. >>>>>> >>>>>> Best Regards, >>>>>> Supreeth >>>>>> >>>>>> >>>>>> On Thu, 14 May 2020 at 17:51, Pavithra Mohanraja < >>>>>> pavimohan3004(a)gmail.com> wrote: >>>>>> >>>>>>> Hi , >>>>>>> >>>>>>> I didn't configure rtpengine ..instead of that only I used >>>>>>> rrpproxy..will it not work? >>>>>>> >>>>>>> If not please tell me how to configure rtpengine and also tell me if >>>>>>> it is really madatory .. why can't we have sip client to sip client rtps >>>>>>> flowing like how it is happening without NAT. >>>>>>> >>>>>>> That's my doubt ..please clarify .. >>>>>>> >>>>>>> Thanks. >>>>>>> >>>>>>> On Thu, May 14, 2020, 8:57 PM supreeth herle < >>>>>>> herlesupreeth(a)gmail.com> wrote: >>>>>>> >>>>>>>> HI, >>>>>>>> >>>>>>>> Hold on a moment, >>>>>>>> >>>>>>>> >>>>>>>>> 10.40.10.3 - pcscf in port 4060 (pcscf with NAT enabled in >>>>>>>>> pcscf.cfg (!#define WITH_NAT) and configured rtpproxy) >>>>>>>>> 10.40.10.12 - icscf in port 4070 >>>>>>>>> 10.40.10.5 - scscf in port 4080 >>>>>>>>> 10.40.10.30 - hss >>>>>>>>> >>>>>>>> >>>>>>>> So i assume you are using the examples provided in the ims folder >>>>>>>> of kamailio repo right? Isn't it using RTPEngine and not RTPProxy? Or have >>>>>>>> you modified the P-CSCF configuration to work with RTPProxy? >>>>>>>> >>>>>>>> Best regards, >>>>>>>> Supreeth >>>>>>>> >>>>>>>> On Thu, 14 May 2020 at 16:39, Pavithra Mohanraja < >>>>>>>> pavimohan3004(a)gmail.com> wrote: >>>>>>>> >>>>>>>>> Hi , >>>>>>>>> I have attached working rtp pcap file. But it is getting reverted >>>>>>>>> back to me . >>>>>>>>> >>>>>>>>> As you said , i have tried restarting rtpproxy several times. but >>>>>>>>> it didnt work . >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> On Thu, May 14, 2020 at 7:11 PM supreeth herle < >>>>>>>>> herlesupreeth(a)gmail.com> wrote: >>>>>>>>> >>>>>>>>>> Thanks for the pcap. Sorry, to ask this but can you also send >>>>>>>>>> pcap of scenario where you see RTP packets? >>>>>>>>>> >>>>>>>>>> Also, did you restart the rtpproxy after altering the >>>>>>>>>> /etc/default/rtpproxy ? >>>>>>>>>> >>>>>>>>>> i.e. >>>>>>>>>> >>>>>>>>>> $ systemctl restart rtpproxy >>>>>>>>>> >>>>>>>>>> Best Regards, >>>>>>>>>> Supreeth >>>>>>>>>> >>>>>>>>>> On Thu, 14 May 2020 at 14:51, Pavithra Mohanraja < >>>>>>>>>> pavimohan3004(a)gmail.com> wrote: >>>>>>>>>> >>>>>>>>>>> Hi Supreeth, >>>>>>>>>>> I have attached pcap files . kindly check and let me know. >>>>>>>>>>> filter of rtp packets and check. >>>>>>>>>>> >>>>>>>>>>> *filter : sip || rtp* >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> On Thu, May 14, 2020 at 5:15 PM supreeth herle < >>>>>>>>>>> herlesupreeth(a)gmail.com> wrote: >>>>>>>>>>> >>>>>>>>>>>> Hi Pavithra, >>>>>>>>>>>> >>>>>>>>>>>> Thanks for the clarification. Can you please attach a pcap of >>>>>>>>>>>> the call where you are not seeing the RTP packets? And, btw is the call >>>>>>>>>>>> getting dropped or something in that scenario? >>>>>>>>>>>> >>>>>>>>>>>> Regards, >>>>>>>>>>>> Supreeth >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> On Thu, 14 May 2020 at 13:34, Pavithra Mohanraja < >>>>>>>>>>>> pavimohan3004(a)gmail.com> wrote: >>>>>>>>>>>> >>>>>>>>>>>>> Hi Supreeth, >>>>>>>>>>>>> >>>>>>>>>>>>> I have kamailio ims installed in 4 vms inside openstack. >>>>>>>>>>>>> >>>>>>>>>>>>> 10.40.10.3 - pcscf in port 4060 (pcscf with NAT enabled in >>>>>>>>>>>>> pcscf.cfg (!#define WITH_NAT) and configured rtpproxy) >>>>>>>>>>>>> 10.40.10.12 - icscf in port 4070 >>>>>>>>>>>>> 10.40.10.5 - scscf in port 4080 >>>>>>>>>>>>> 10.40.10.30 - hss >>>>>>>>>>>>> >>>>>>>>>>>>> so all these vms are running in same subnet. >>>>>>>>>>>>> I have a floating ip associated to pcscf. >>>>>>>>>>>>> Now I have two zoiper clients outside openstack >>>>>>>>>>>>> zoiper1 - 10.0.2.15 (NAT Enabled Network adaptor) >>>>>>>>>>>>> zoiper2 - 10.0.2.15 >>>>>>>>>>>>> >>>>>>>>>>>>> when i am making a call from zoiper1 to zoiper2, call is going >>>>>>>>>>>>> properly . SIP messages are coming properly in wireshark packets. >>>>>>>>>>>>> when i filter for RTP packets in wireshark, it is not coming. >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> when i have the zoiper clients inside openstack itself ,i can >>>>>>>>>>>>> able to see RTP Packets in wireshark between two sip clients without any >>>>>>>>>>>>> NAT related configurations done. >>>>>>>>>>>>> >>>>>>>>>>>>> Kindly help me. >>>>>>>>>>>>> Thanks. >>>>>>>>>>>>> >>>>>>>>>>>>> On Thu, May 14, 2020 at 4:49 PM supreeth herle < >>>>>>>>>>>>> herlesupreeth(a)gmail.com> wrote: >>>>>>>>>>>>> >>>>>>>>>>>>>> Can you explain some more when you mean by "RTP alone is not >>>>>>>>>>>>>> coming."? And, how are you checking/verifying that? >>>>>>>>>>>>>> >>>>>>>>>>>>>> Regards, >>>>>>>>>>>>>> Supreeth >>>>>>>>>>>>>> >>>>>>>>>>>>>> On Thu, 14 May 2020 at 13:12, Pavithra Mohanraja < >>>>>>>>>>>>>> pavimohan3004(a)gmail.com> wrote: >>>>>>>>>>>>>> >>>>>>>>>>>>>>> Hi , >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Yes I have configured as you mentioned only . In my >>>>>>>>>>>>>>> pcscf only i have rtpproxy also configured. >>>>>>>>>>>>>>> Floating IP i have given in listen address also as >>>>>>>>>>>>>>> listen = 10.40.10.3:4060 advertise 10.45.4.22:4060 >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> 10.40.10.3 - private ip >>>>>>>>>>>>>>> 10.45.4.22 - floating ip >>>>>>>>>>>>>>> Still RTP alone is not coming. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Is it the only way of configuring NAT ? >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Thanks. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> On Thu, May 14, 2020 at 4:37 PM supreeth herle < >>>>>>>>>>>>>>> herlesupreeth(a)gmail.com> wrote: >>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Hello Pavithra, >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Have you tried advertising the floating IP of the VM >>>>>>>>>>>>>>>> running RTPProxy as follows? >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Edit /etc/default/rtpproxy file as follows: >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> # Additional options that are passed to the daemon. >>>>>>>>>>>>>>>> EXTRA_OPTS="-l 172.24.15.30 -d DBUG:LOG_LOCAL0" >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> here, "-l <PUBLIC_IP>" >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Hope it helps. >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Regards, >>>>>>>>>>>>>>>> Supreeth >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> On Thu, 14 May 2020 at 13:01, Pavithra Mohanraja < >>>>>>>>>>>>>>>> pavimohan3004(a)gmail.com> wrote: >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Hi , >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Could anybody please give me the approach how to configure >>>>>>>>>>>>>>>>> NAT in kamailio IMS. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> I have installed kamailio IMS in openstack environment >>>>>>>>>>>>>>>>> which has private ips for all four vms (each component in one vm). >>>>>>>>>>>>>>>>> pcscf has one floating IP attached. I have enabled "WITH_NAT" and >>>>>>>>>>>>>>>>> configured rtpproxy . >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> when i make a call from outside of openstack using 2 >>>>>>>>>>>>>>>>> zoiper clients with outbound proxy keeping floating ip of pcscf, I am >>>>>>>>>>>>>>>>> getting 200 ok with all SIP packets correctly flowing but there is no RTP >>>>>>>>>>>>>>>>> Packet. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> When i try the same within openstack environment without >>>>>>>>>>>>>>>>> using public ip, I am getting RTP Packets. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Kindly tell me the procedure of NAT Enablement or do i >>>>>>>>>>>>>>>>> need to do anything other than this. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Kindly help because i am stuck with this for long time. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Thanks. >>>>>>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>>>>>> 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 >>>>>>>>>>>>>>> >>>>>>>>>>>>>> _______________________________________________ >>>>>>>>>>>>>> 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 >>>>>>>>>>>> >>>>>>>>>>> _______________________________________________ >>>>>>>>>>> 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 >>>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> 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 >>>>>> >>>>> [View Less]

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Re: [SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag)
by Martin W Woscek 24 Jun '20

24 Jun '20

Hi, The issue is the P should be sending to the UE not back to the I like the failed flow: Asterisk16-[phone] --->ICSCF--->S-CSCF--->P-CSCF--->I-CSCF ---> [604 HSS user unknown] to Asterisk16[phone] The successful call should look like: Asterisk16-[phone] --->ICSCF--->S-CSCF--->P-CSCF--->Kamailio-[UE] Does anyone have a example P-cscf scripts including the mt.cfg or other script file that make this work? The reverse calls work just fine. Thanks, _Martin From: … [View More]Daniel-Constantin Mierla <miconda(a)gmail.com> Sent: Wednesday, June 24, 2020 4:53 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>; Martin W Woscek <mwoscek(a)mitre.org> Subject: [EXT] Re: [SR-Users] Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag) Hello, Kamailio is not generating the 200ok for INVITE (calls), it just sends out what it received from end point, after consuming own Via header (of course, other headers can be changed based on config rules, but Kamailio is not in charge of setting To-tag). Cheers, Daniel On 19.06.20 21:19, Martin W Woscek wrote: Hello all, SIP/UE (boghe or imsdroid) client registered to Kamailio makes call to an Asterisk registered SIP phone is successful: [UE]-Kamailio--->INVITE--->Asterisk16-[phone] [UE]-Kamailio--->200 OK with to_tag--->Asterisk16-[phone] But in reverse direction for a call, Kamailio does not return the SIP OK so no to_tag is sent so call fails to ring and complete: [phone]-Asterisk16 --->INVITE--->Kamailio Where the UE device never receives the INVITE, Asterisk never gets and 200 OK message with the to_tag from the I-CSCF, the call flow itself gets lost in Kamailio, where the P-CSCF sends final INVITE to I-CSCF and and ultimately a 604 HSS user unknown message is sent back to Asterisk from the I-CSCF. Basically Im using the default "sample" configs for both the P and the I-CSCF. Our sauce is in the S-CSCF for out going calls that originate by a registered UE. Any insight or sample Kamailio configuration that Im lacking? Has anyone done this and could share the asterisk and Kamailio script snippets that make it possible. 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 -- Daniel-Constantin Mierla -- www.asipto.com<http://www.asipto.com> www.twitter.com/miconda<http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda> Funding: https://www.paypal.me/dcmierla [View Less]

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Kamailio IMS using returned SRV records for transport slelection
by BALL SUN 24 Jun '20

24 Jun '20

Hi All I am new to Kamailio IMS, and I just setup a testing environment where having P-CSCF, I-CSCF and S-CSCF. I would like to use make below works 1. SIP client <-> P-CSCF (either UDP or TCP, based on the transport tag) 2. P-CSCF <-> I-CSCF (TCP by default) 3. I-CSCF <-> S-CSCF (TCP by default) For point 2, I observed that P-CSCF actually will send DNS SRV query to DNS server for resolving _sip._udp. and _sip._tcp. I believed we can just disable the "_sip._udp." so that P-… [View More]

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SECFILTER reload from Siremis
by Social Boh 24 Jun '20

24 Jun '20

Hello list, is posible to reload SECFILTER data directly from Siremis? or the only way is: kamcmd secfilter.reload Regards -- --- I'm SoCIaL, MayBe

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dialog's track_cseq_updates seems to break uac_auth() in this configuration
by George Diamantopoulos 24 Jun '20

24 Jun '20

Hello again, This might turn out to be a bug, but I'm posting here first because my config is fairly complicated and it might not be the sanest it could be. The problem manifests as follows: with the sole change being the enabling of track_cseq_updates for the dialog module, kamailio fails to process the 100 Trying it receives following authentication initiated by the uac_auth() function. Without the track_cseq_updates option enabled, the call proceeds successfully, albeit with the CSeq for … [View More]the auth-carrying INVITE having the same value as the original. Here's a simplified version of the configuration routes involved in this: request_route { ... route(DISPATCH_PROVIDER); ... } route[DISPATCH_PROVIDER] { if ( !ds_select_dst("10", "8") ) { t_send_reply("503", "Downstream carrier unavailable"); exit; } t_on_branch_failure("DISPATCH_PROVIDER_FAILOVER"); t_on_branch("PROVIDER_BRANCH"); t_on_failure("DISPATCH_PROVIDER_FAILURE"); route(RELAY); exit; } failure_route[DISPATCH_PROVIDER_FAILURE] { if (t_is_canceled()) { exit; } if ( t_check_status("401|407") ) { $avp(arealm) = "authrealm.com"; $avp(auser) = "authusername"; $avp(apass) = "verys3kr1t"; uac_auth(); t_on_branch("PROVIDER_BRANCH"); t_on_branch_failure("DISPATCH_PROVIDER_FAILOVER"); t_on_failure("DISPATCH_PROVIDER_FAILURE"); $du = "sip:" + $T_rpl($si) + ":" + $T_rpl($sp); // I'll explain why this is here later in this e-mail t_relay(); exit; } } branch_route[PROVIDER_BRANCH] { uac_replace_from("$dlg_var(my_new_from)"); uac_replace_to("$dlg_var(my_new_to)"); $rU = "my new RURI user"; } event_route[tm:branch-failure:DISPATCH_PROVIDER_FAILOVER] { if (t_is_canceled()) { exit; } if ( t_check_status("401|407") ) { return; # next DST - only for 5xx or local timeout } else if ( t_check_status("5[0-9][0-9]") || (t_branch_timeout() && !t_branch_replied()) ) { if ( ds_next_dst() ) { t_on_branch("PROVIDER_BRANCH"); t_on_branch_failure("DISPATCH_PROVIDER_FAILOVER"); t_on_failure("DISPATCH_PROVIDER_FAILURE"); route(RELAY); exit; } else { xlog("L_NOTICE", "--- SCRIPT_DISPATCH_PROVIDER_FAILOVER: Failed to route request to PROVIDER! Giving Up. Negative response will be sent upstream.\n"); } } } One will rightfully wonder why use both branch-failure routes and failure_route to handle things. Well, it's not obvious from this because the relevant parts are removed for brevity, but I'm generally using branch failure event routes to try other destinations in the same destination set, and failure routes to additionally try other downstream "providers" in case all entries is the current destination set have failed. I don't think these actions/routes are relevant because the problem manifests without any of these failover mechanisms (of switching over to the next provider) engaging. Regarding the line "$du = "sip:" + $T_rpl($si) + ":" + $T_rpl($sp);" in the failure_route, which might seem a little odd, this was added because without it, uac_auth() will send the auth-carrying INVITE always to the first destination in the destination set even if ds_next_dst has been called from the branch-failure event route. Unfortunately I haven't been able to determine if this actually helped, because I have only received a 5xx error once from the downstream peer before the "fix" and haven't been able to reproduce this since then. But that's another issue unrelated to the one I'm asking about here. So what happens with track_cseq_updates enabled is, after uac_auth() successfully sends the authenticated INVITE and the peer starts sending provisional responses, kamailio doesn't seem to acknowledge them. Instead, it will retransmit the auth INVITE as if there was a firewall preventing the 1xx responses from being admitted. Then, the timeout will engage and the configuration script will behave as if t_branch_timeout returns true (it will do ds_next_dst). So I was wondering if this is to be expected with this configuration, or if this should be reported as a bug. Thanks! Best regards, George [View Less]

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Kamailio-5.3.3. NAT and RTP not working
by Pavithra Mohanraja 24 Jun '20

24 Jun '20

Hi , Could anybody please give me the approach how to configure NAT in kamailio IMS. I have installed kamailio IMS in openstack environment which has private ips for all four vms (each component in one vm). pcscf has one floating IP attached. I have enabled "WITH_NAT" and configured rtpproxy . when i make a call from outside of openstack using 2 zoiper clients with outbound proxy keeping floating ip of pcscf, I am getting 200 ok with all SIP packets correctly flowing but there is no RTP … [View More]

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multiple domains with microsoft team
by Leanderson Correia 24 Jun '20

24 Jun '20

I'm testing here to use multiple domains with microsoft teams and I'm having trouble making and receiving calls. It seems that when connecting a call, kamailio keeps the tls connection active and reuses it, making the domain parameters used the wrong ones for new call. Can someone help me, or give a tip? -- Att, Lean

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Presence and call queueing - Kamailio and Asterisk
by Pover A.V. 24 Jun '20

24 Jun '20

Summary: Using Kamailio to load balance between Asterisk servers. Trouble getting call queuing working because can't get device state hints to work reliably. - Hello I'd be grateful for any advice that can be offered for this problem to do with presence and call queueing. I have 3 Kamailio (v5.3.4) servers that load-balance incoming traffic from various sources: a legacy PBX, soft phone clients, and a SIP trunking service. Sitting behind the 3 Kamailio servers are 2 Asterisk (v16 LTS)… [View More] servers. Kamailio acts as a registrar and user location service. Asterisk is routing calls and (should be) providing device state information for each extension. An external script builds a list of PJSIP endpoints based on the content of the Kamailio subscriber table, and deploys it to the Asterisk servers whenever a change is detected. The AOR for each record is configured to point to Kamailio. No authentication details are attached to these endpoints, as Kamailio is doing authentication before dispatching a request to Asterisk. Whenever Kamailio receives an INVITE, it passes this to an Asterisk server using the dispatcher module (round-robin algorithm '4'); asterisk then processes the request. At the moment, everything is working except for presence/device states and call queueing. I've tried to set up 'presence', 'pua', 'dialog' and '(presence/pua)_dialoginfo' modules on Kamailio with limited success. The presence state when a user toggles 'do not disturb' on their soft phone is passed to other users who are subscribed (watching), however when a user makes a call, no presence updates are seen by subscribed users (the extensions show as 'Idle' or 'Available' instead of 'In Use' or 'Busy'). Asterisk is able to report that an extension is 'Ringing' or 'InUse' when a call is made to that extension, because an INVITE to an internal extension matches a pattern in the dialplan and Asterisk routes it to a PJSIP endpoint with the same name. However, when somebody makes a call, Asterisk is unable to track the state of the extension making the call because the call is going via Kamailio so Asterisk is not seeing the caller as an internal extension. We need a way to make Asterisk think the call is coming from an internal device if the caller ID matches a PJSIP endpoint. Both Asterisk servers are configured to share device state updates with each other, and this is working. Without being able to see device state for an internal calling party, we will be unable to reliably set up call queues in Asterisk. Using the 'mohqueue' module on Kamailio is not something we want to do because of its reliance on a database, which is creating a single point of failure - if the database server is down for any reason, Kamailio refuses to start. Ideally, we would achieve some form of distributed call queueing, however the simplest option seems to be setting up a queue on one Asterisk server and having an active-passive high availability configuration, should a server go down. I'm looking for advice on how best to keep track of what is happening in a call, and for Asterisk to know the states of each extension and make these available in a dialplan hint. Or any other suggestions that might help to achieve call queueing in this setup. Thank you [View Less]

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Dialog Profiles
by Ilie Soltanici 24 Jun '20

24 Jun '20

Hello, Is there any way in Kamailio to get the profile values the current dialog is part of? For example, I have one dialog with the following profiles: profiles: { { test: abcd } test: 1234 } I need in the reply route to get the values for the test dialog profile. (abcd, 1234). I know I can use get_profile_size, but that will give the number of profiles of the current dialog, but I need the values to loop through. Thanks.

2 2

Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag)
by Martin W Woscek 24 Jun '20

24 Jun '20

Hello all, SIP/UE (boghe or imsdroid) client registered to Kamailio makes call to an Asterisk registered SIP phone is successful: [UE]-Kamailio--->INVITE--->Asterisk16-[phone] [UE]-Kamailio--->200 OK with to_tag--->Asterisk16-[phone] But in reverse direction for a call, Kamailio does not return the SIP OK so no to_tag is sent so call fails to ring and complete: [phone]-Asterisk16 --->INVITE--->Kamailio Where the UE device never receives the INVITE, Asterisk never gets and … [View More]

2 1

Kamailio ims video calls
by Pavithra Mohanraja 24 Jun '20

24 Jun '20

Hi, Could anyone please give me some idea on how video call works in IMS. Does it need rtp proxy to work? Or is it the same way how voice call works ? It would be good , if anybody share some links on video calls as well voice conferencing. Thanks, pavithra

2 1

kamailio 5.2.2 enabling hashing_type
by Martin W Woscek 24 Jun '20

24 Jun '20

Hello, I'm revisiting some initial issues I had with P-CSCF. I could not set the hashing type in my PCSCF (kamalio 5.2.2) so I simply commented it out and moved on. But now, I may need it as the P-CSCF is not handling INVITES coming from I-CSCF that originated from external SIP server, where the endpoint (UE) is registered to the IMS but the P-CSCF sends the INVITE back to the S rather that the registered UE and fails. kamailio.cfg: modparam("ims_usrloc_pcscf", "hashing_type", 1) So with … [View More]

2 1

kamailio5.2.2 calling t_relay(host, port)
by Martin W Woscek 24 Jun '20

24 Jun '20

Hello all, Im trying to call t_relay() with host and port as documented in: https://www.kamailio.org/docs/modules/5.2.x/modules/tm.html However, I get a subst_str(): no match on P-CSCF start and therefore fails. Is t_relay(host, port) supported as documented? Am I missing something? Thanks, _Martin

2 1

Expired transaction
by Marat Gareev 24 Jun '20

24 Jun '20

Hello! I'm trying to manually reproduce expired transactions which couldn't be cleared. I had one idea, but it was unsuccessful. I tried to send SIGSTOP to a group of Kamailio processes while the transaction is live. The idea is to freeze processes for a while, then continue to work, so that by this time the transaction deletion time was missed. But after a few seconds, the transaction is still cleared. Do you have any ideas for reproduction? P.S. I'm sorry If you received my incoherent … [View More]

2 1

ERROR: pua [send_publish.c:304]: publ_cback_func(): No Expires header found
by Denys Pozniak 23 Jun '20

23 Jun '20

Hello! Could someone point me to the possible source of the problem? Jun 16 09:51:06 proxy-edge /usr/sbin/kamailio[9645]: ERROR: pua [send_publish.c:304]: publ_cback_func(): No Expires header found Jun 16 09:51:07 proxy-edge /usr/sbin/kamailio[9640]: ERROR: pua [send_publish.c:304]: publ_cback_func(): No Expires header found Jun 16 09:51:07 proxy-edge /usr/sbin/kamailio[9600]: ERROR: pua [send_publish.c:304]: publ_cback_func(): No Expires header found Jun 16 09:51:07 proxy-edge /usr/sbin/… [View More]kamailio[9660]: ERROR: pua [send_publish.c:304]: publ_cback_func(): No Expires header found Modules settings: *loadmodule "pua.so"* *loadmodule "pua_dialoginfo"* *loadmodule "dialog.so" * *modparam("pua", "default_expires", 1200)* *modparam("pua", "outbound_proxy", LOCAL_IP_ADDR)* *modparam("pua_dialoginfo", "include_callid", 1)* *modparam("pua_dialoginfo", "include_localremote", 1)* *modparam("pua_dialoginfo", "include_tags", 1)* *modparam("pua_dialoginfo", "send_publish_flag", 8)* *modparam("dialog", "dlg_flag", 4)modparam("dialog", "db_mode", 0)modparam("dialog", "enable_stats", 1)modparam("dialog", "dlg_match_mode", 1)modparam("dialog", "default_timeout", 7200) modparam("dialog", "hash_size", 8192)* *request_route{* *...* *setflag(8);* *...* *}* Generated SIP PUBLISH: *PUBLISH sip:extension@sip.company.com <sip%3Aextension(a)sip.company.com> SIP/2.0* *Via: SIP/2.0/UDP 8.8.8.8;branch=z9hG4bK3e55.afdce977000000000000000000000000.0* *To: <sip:extension@sip.company.com <sip%3Aextension(a)sip.company.com>>* *From: <sip:extension@sip.company.com <sip%3Aextension(a)sip.company.com>>;tag=e98d9fc94bada9746d67946ab287a76f-ba77* *CSeq: 10 PUBLISH* *Call-ID: 72ef85f374c8e703-9659(a)10.10.10.10 <72ef85f374c8e703-9659(a)10.10.10.10>* *Content-Length: 578* *User-Agent: proxy* *Max-Forwards: 70* *Event: dialog* *Expires: 7201* *Content-Type: application/dialog-info+xml* *<?xml version="1.0"?>* *<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:extension@sip.company.com <sip%3Aextension(a)sip.company.com>">* * <dialog id="162092997-5060-464(a)BJC.BGI.CA.BBI" call-id="162092997-5060-464(a)BJC.BGI.CA.BBI" direction="initiator">* * <state>Trying</state>* * <remote>* * <identity>sip:1234567890@sip.company.com <sip%3A1234567890(a)sip.company.com></identity>* * <target uri="sip:1234567890@sip.company.com <sip%3A1234567890(a)sip.company.com>"/>* * </remote>* * <local>* * <identity>sip:extension@sip.company.com <sip%3Aextension(a)sip.company.com></identity>* * <target uri="sip:extension@sip.company.com <sip%3Aextension(a)sip.company.com>"/>* * </local>* * </dialog>* *</dialog-info>* -- BR, Denys Pozniak [View Less]

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icscf start error
by Pavithra Mohanraja 22 Jun '20

22 Jun '20

Hi, I have installed kamailio - 5.3.4 version where i am facing below issues in icscf while starting kamailio. db_mysql is loaded but .. what could be the issue? icscf : Aliases: *: ims.sip.example.com:* 0(789) INFO: <core> [core/tcp_main.c:5042]: init_tcp(): using epoll_lt as the io watch method (auto detected) 0(791) INFO: pike [pike.c:97]: pike_init(): PIKE - initializing 0(791) INFO: pike [ip_tree.c:81]: init_lock_set(): probing 256 set size *0(791) ERROR: <… [View More]

2 3

Kamailio v5.3.5 Released
by Daniel-Constantin Mierla 22 Jun '20

22 Jun '20

Hello, Kamailio SIP Server v5.3.5 stable release is out. This is a maintenance release of the latest stable branch, 5.3, that includes fixes since the release of v5.3.4. 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 v5.3.5. For more details about version 5.3.5 (including links and guidelines to download the tarball or … [View More]

1 0

reason=404 (Not Found - destination user not found on this S-CSCF)
by Pavithra Mohanraja 22 Jun '20

22 Jun '20

Hi, I am using kamailio ims-5.3.4 where i am facing this kind of problem during route(term). call registration is happening successfully. In call Establishment , during route(term) , I am facing destination user not found . what could be the problem?. can anyone please help me. Thanks, Pavithra

2 2

Releasing Kamailio v5.3.5
by Daniel-Constantin Mierla 22 Jun '20

22 Jun '20

Hello, I am considering to release next week (likely on Monday, June 22) the next minor version out of branch 5.3, respectively 5.3.5. Should anyone be aware of issues not reported to the bug tracker or commits for fixes in master not backported to 5.3 branch, let us know. For new issues, the best is to report them on the tracker. For commits, you can provide links/hashids or make pull requests. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 1

Re: [SR-Users] 403 domain not served error in scscf
by David Villasmil 20 Jun '20

20 Jun '20

you shouldn't write to me directly. You can write to the list and address me. Anyway, kamailio thinks the message received is NOT directed to it. You should add an "alias" with the IP you're sending to (or the FQDN) Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 On Sat, Jun 20, 2020 at 9:01 PM Pavithra Mohanraja <pavimohan3004(a)gmail.com> wrote: > Hi David, > Could you please help me in this issue > > ---------- Forwarded message --… [View More]

2 1

How do I test the Kamailio service ？
by zhaosl＠shinetechchina.com 20 Jun '20

20 Jun '20

Hi : Now I set up a Kamailio Service on the server, it works now. But I didn`t know how to test the server by trying the voice tool.Can you give me some advices , tools or methods ? Thanks. zhaosl(a)shinetechchina.com

3 2

Re: [SR-Users] R: R: R: R: R: acc time mode 2 wrong timestamp
by mohsen khashei 20 Jun '20

20 Jun '20

Hi can you share your kamaili config. I want to add some extra columns to cdr table like what you did. Thanks. Sent from Mail<https://go.microsoft.com/fwlink/?LinkId=550986> for Windows 10

2 1

Re: [SR-Users] R: R: R: R: R: acc time mode 2 wrong timestamp
by mohsen khashei 20 Jun '20

20 Jun '20

1 0

403 domain not served error in scscf
by Pavithra Mohanraja 19 Jun '20

19 Jun '20

Hi, I am using kamailio IMS - 5.3.4 when I am making a call between two clients using zoiper , Call establishment is done successfully . But during BYE packet sent from caller to callee , it is reaching till S-CSCF and giving 403 Domain not served .. Kindly help me in this regard. I have sent lot of mails regarding this error and no replies .. Kindly help .. Error in scscf : Jun 19 13:29:17 ubuntu-VirtualBox kamailio[21891]: DEBUG: <core> [core/mem/q_malloc.c:526]: qm_free(): freeing … [View More]

1 0

Development for v5.4.x is frozen
by Daniel-Constantin Mierla 19 Jun '20

19 Jun '20

Hello, the formal notification that the development for the next major version 5.4.0 is now frozen. The focus has to be on testing the master branch. Also, the master branch should not get commits with new features till the branch 5.4 is created, expected to happen in 2-4 weeks, a matter of how testing goes on. Meanwhile, the commits with new features in the C code can be pushed to personal branches, pull requests can still be done, but they will be merged after branching 5.4. Can still be … [View More]

1 0

MD5 and SHA-256 instead of MD5 or SHA-256...
by Aymeric Moizard 18 Jun '20

18 Jun '20

Hi All, I'd like to improve my setup by switching to SHA-256. However, as a first step, I would like to offer both MD5 and SHA-256 in 2 different WWW-Authenticate header. If I'm correct, this is not doable with the latest auth module? Is this a planned feature? As an alternative, I would like to decide the algorithm in the script instead of a module parameter. It looks to me this is also not doable? Again, is this a planned feature? Thanks to all, Regards Aymeric -- Antisip - http://www.antisip.com

5 10

Freezing development for Kamailio v5.4.0
by Daniel-Constantin Mierla 18 Jun '20

18 Jun '20

Hello, based on the discussions during the last online devel meeting, the plan is to release the next major version (Kamailio v5.4.0) around the mid of this summer. Therefore I propose to freeze the development on Thursday, June 18, 2020. Then we will have about 4-6 weeks of testing before the release is out. Master will be unfrozen when the branch 5.4 is created (usually that was done after 3-4 weeks in the testing phase). If you are planning to introduce new features in the 5.4 release … [View More]

2 4

removing xavp from the stack.
by David Villasmil 18 Jun '20

18 Jun '20

Hello guys, if i do something like: $(xavp(my_var)) = "1"; $(xavp(my_var)) = "2"; $(xavp(my_var)) = "3"; it is store like: $(xavp(my_var)) = "1"; -> index 2 $(xavp(my_var)) = "2"; -> index 1 $(xavp(my_var)) = "3"; -> index 0 and later i need to use and remove from the stack, like this: use and remove: $(xavp(my_var)) --> would remove index 0 Then the stack would remain like: $(xavp(my_var)) = "1"; -> index 1 $(xavp(my_var)) = "2"; -> index 0 Simply popping the index 0 … [View More]

2 1

dispatcher seems to use the default client certificate
by Mack Hendricks 18 Jun '20

18 Jun '20

Hey All, I'm attempting to use dispatcher to send probe messages using TLS for two different domains. I'm providing the socket attribute, which maps to a certificate in /etc/kamailio/tls.cfg. But, it seems to always select the default client cert, which is not the certificate I want to use. My attrs column in dispatcher looks like this: socket=tls:142.93.159.231:5061;ping_from=sip:mack.dopensource.com socket=tls:142.93.159.231:5062;ping_from=sip:levin.dopensource.com Is there some way … [View More]

5 5

Remove a Line from SDP
by egemen ulus 18 Jun '20

18 Jun '20

Hi all, I am trying to remove a line from the SDP body of INVITE on kamailio 4.4 via SDPOPS module but it does not work even though there is no any error message. This is how I configure kamailio.cfg, it matches with if condition and executes "sdp_remove_line_by_prefix" function, but when I check sended INVITE, I still see the line that beginning with "a=rtpmap". I thought, in the RELAY function, the sdp body might not prepared yet? Is there any suggestion to remove a line from the SDP body? … [View More]

2 2

null branch(count) variable
by Ilie Soltanici 18 Jun '20

18 Jun '20

Hello, As per documentation after lookup() function I have to get the number of branches from the main branch by using $branch(count) variable, but for some reason, I'm getting always only <null> value, can someone help me to understand what I'm doing wrong? lookup(); xlog("Branch Count: $branch(count)\n"); Kamailio Version: 5.2.7 Thanks.

3 4

403 domain not served error in IMS kamailio
by Pavithra Mohanraja 18 Jun '20

18 Jun '20

Hi, Could any one please help me in this issue ASAP. This is logs from SCSCF. Jun 18 06:28:02 396779d9d4e7 kamailio[1796]: DEBUG: <core> [core/socket_info.c:630]: grep_sock_info(): checking if host==us: 10==10 && [172.17.0.8] == [172.17.0.5] Jun 18 06:28:02 396779d9d4e7 kamailio[1796]: DEBUG: <core> [core/socket_info.c:637]: grep_sock_info(): checking if port 4080 (advertise 0) matches port 5060 Jun 18 06:28:02 396779d9d4e7 kamailio[1796]: DEBUG: <core> [core/… [View More]

1 0

docker images for kamailio IMS
by Pavithra Mohanraja 18 Jun '20

18 Jun '20

Hi , Could anyone kindly tell me if there is docker images for kamailio IMS. I can able to see docker images for only Kamailio as a SIP server. Please help. Thanks, Pavithra

4 10

Re: [SR-Users] kamailio 5.2.2 enabling hashing_type -Please Ignore
by Martin W Woscek 17 Jun '20

17 Jun '20

Found my answer, use correct version of doc: https://kamailio.org/docs/modules/5.2.x/modules/ims_usrloc_pcscf.html#idm10… From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Martin W Woscek Sent: Wednesday, June 17, 2020 8:47 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: [SR-Users] kamailio 5.2.2 enabling hashing_type Hello, I'm revisiting some initial issues I had with P-CSCF. I could not set the hashing type in my PCSCF (… [View More]

1 0

Kamailio IMS support
by Pavithra Mohanraja 16 Jun '20

16 Jun '20

Hi, I am trying to connect EPC with IMS. EPC is separate installation of comac-in-a-box . IMS is kamailio based. Do i have SGi interface configurations in kamailio so that i can establish communication between EPC and IMS. One more clarification. I have seen PCRF communication can be done using CDP .Is there any way to build PCRF as part of kamailio. Kindly help .. Thanks, Pavithra

2 1

Re: [SR-Users] [EXT] RE: kamailio 522 S-CSCF failing to start
by Martin W Woscek 16 Jun '20

16 Jun '20

Hi Henning, The following are the mysql related logging I see on initial spinup of the Kamailio 5.2.2 IMS VM: Jun 16 06:15:28 ubuntu systemd[1]: Starting MySQL Community Server... Jun 16 06:15:33 ubuntu systemd[1]: Started MySQL Community Server. Jun 16 06:16:09 ubuntu /usr/local/sbin/kamailio[1617]: ERROR: db_mysql [km_dbase.c:127]: db_mysql_submit_query(): driver error on query: Table 'scscf.impu_subscriber' doesn't exist (1146) Jun 16 06:16:09 ubuntu kernel: traps: kamailio[1617] general … [View More]protection fault ip:7f3dfa2874f0 sp:7ffe387dc8c0 error:0 in libmysqlclient.so.20.3.16[7f3dfa261000+33c000] Our mysql version: mysql --version mysql Ver 14.14 Distrib 5.7.29, for Linux (x86_64) using EditLine wrapper Thanks, _Martin From: Henning Westerholt <hw(a)skalatan.de> Sent: Monday, June 8, 2020 4:50 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Cc: Martin W Woscek <mwoscek(a)mitre.org> Subject: [EXT] RE: kamailio 522 S-CSCF failing to start Hello, the error message below is from the DB startup version check. So do you have sometimes DB errors on this virtual machine? Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/> From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of Martin W Woscek Sent: Monday, June 8, 2020 2:46 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> Subject: [SR-Users] kamailio 522 S-CSCF failing to start Hello, There are times that our S-CSCF module just wont start. I’ve tried correlating the issue with some operation such as not de-registering an IMS client (Boghe or imsdroid) prior to shutting down the Kamailio server. But this isn’t always the case. Being that we are running this an a Ubuntu Virtual Machine, simply overwriting the image with a backed up VM is the temporary solution. I’ve dropped and created the db for S (kamdbctl) and rearranged the order of loading the presence module with same result. Status of the S-CSCF daemon on failure to start: # systemctl status kamailio_scscf ● kamailio_scscf.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio_scscf; generated) Active: inactive (dead) Docs: man:systemd-sysv-generator(8) Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jun 08 08:22:55 ubuntu kamailio_scscf[16495]: * already running Jun 08 08:22:55 ubuntu kamailio_scscf[16495]: ...done. Jun 08 08:22:55 ubuntu systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Jun 08 08:24:03 ubuntu systemd[1]: Stopping LSB: Start the Kamailio SIP proxy server... Jun 08 08:24:03 ubuntu kamailio_scscf[16517]: * Stopping Kamailio IMS S-CSCF: kamailio_scscf Jun 08 08:24:03 ubuntu kamailio_scscf[16517]: ...done. Jun 08 08:24:03 ubuntu systemd[1]: Stopped LSB: Start the Kamailio SIP proxy server. root@ubuntu:~/IMS_manual_scripts# systemctl start kamailio_scscf root@ubuntu:~/IMS_manual_scripts# systemctl status kamailio_scscf ● kamailio_scscf.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio_scscf; generated) Active: active (exited) since Mon 2020-06-08 08:27:23 EDT; 58s ago Docs: man:systemd-sysv-generator(8) Process: 16543 ExecStart=/etc/init.d/kamailio_scscf start (code=exited, status=0/SUCCESS) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: <core> [db.c:457]: db_check_table_version(): invalid version 0 for table presentity found, expected 5 (check table structure and ta Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: presence [presence.c:367]: mod_init(): error during table version check Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: <core> [core/sr_module.c:898]: init_mod(): Error while initializing module presence (/usr/local/lib64/kamailio/modules/presence.so) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: WARNING: <core> [core/mem/q_malloc.c:480]: qm_free(): WARNING: free(0) called from cdp_avp: cdp_avp_mod.c: cdp_avp_destroy(226) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jun 08 08:27:23 ubuntu kamailio_scscf[16543]: * already running Jun 08 08:27:23 ubuntu kamailio_scscf[16543]: ...done. Jun 08 08:27:23 ubuntu systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Any insight is appreciated. Thanks, _Martin [View Less]

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I use kamailio with fusionpbx And I have a question
by idan katz 16 Jun '20

16 Jun '20

HI I use kamailio with fusionpbx The fusionpbx on one server with a fixed IP address The kamailio on one server with a fixed IP address Two separate servers I use domains I redirect the domains to the kamailio server To register for phones Everything is fine and it works fine But I'm trying to connect to the fusionpbx GUI In a specific domain I'm not going into the fusionpbx The domain is targeted to kamailio Because of the telephone registration I need to provide a … [View More]

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Re: [SR-Users] How to resolve these mod_param errors in Kamailio 5.3.1?
by Henning Westerholt 15 Jun '20

15 Jun '20

Hello, please keep the list in CC have a look to the docs later in the README: 3.13. xavp_dst_mode (int) Control what fields are added to the XAVP specified by xavp_dst parameter. Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/> From: Muhammad Maab Saleem <Maab.Saleem(a)telenor.com.pk> Sent: Monday, June 15, 2020 7:10 AM To: Henning Westerholt <hw(a)skalatan.de> Subject: RE: How to resolve these … [View More]mod_param errors in Kamailio 5.3.1? Hi, Henning. Thanks for your response. However, as I pointed out to him, this only fixes one of the parameter. I have three others where there is a conflict. I can't find any replacements for them. Do you have any pointers regarding them? Best, Maab Sensitivity: Internal From: Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> Sent: Saturday, June 13, 2020 6:50 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> Cc: Muhammad Maab Saleem <Maab.Saleem(a)telenor.com.pk<mailto:Maab.Saleem@telenor.com.pk>> Subject: RE: How to resolve these mod_param errors in Kamailio 5.3.1? Hello, To quote the e-mail from Ovidiu some days ago: "Take a look at the new param: https://kamailio.org/docs/modules/5.3.x/modules/dispatcher.html#dispatcher.…<https://eur02.safelinks.protection.outlook.com/?url=https%3A%2F%2Fkamailio.…> Up to version 5.2, regular AVPs were used, but starting with version 5.3 the regular AVPs were replaced by an XAVP." So you need to replace the old parameter with the new one. Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/<https://eur02.safelinks.protection.outlook.com/?url=https%3A%2F%2Fskalatan.…> Kamailio services - https://gilawa.com<https://eur02.safelinks.protection.outlook.com/?url=https%3A%2F%2Fgilawa.co…> From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of Muhammad Maab Saleem Sent: Friday, June 12, 2020 8:37 AM To: sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Subject: [SR-Users] How to resolve these mod_param errors in Kamailio 5.3.1? This is the error I get when I run ./kamailio -f /etc/kamailio/kamailio.cfg -E ________________________________ loading modules under config path: /usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/:/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/ 0(28369) ERROR: [core/modparam.c:140]: set_mod_param_regex(): parameter <dst_avp> of type <1> not found in module 0(28369) CRITICAL: [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 378, column 61: Can't set module parameter 0(28369) ERROR: [core/modparam.c:140]: set_mod_param_regex(): parameter <grp_avp> of type <1> not found in module 0(28369) CRITICAL: [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 379, column 61: Can't set module parameter 0(28369) ERROR: [core/modparam.c:140]: set_mod_param_regex(): parameter <cnt_avp> of type <1> not found in module 0(28369) CRITICAL: [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 380, column 61: Can't set module parameter 0(28369) ERROR: [core/modparam.c:140]: set_mod_param_regex(): parameter <sock_avp> of type <1> not found in module 0(28369) CRITICAL: [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 381, column 63: Can't set module parameter ERROR: bad config file (4 errors) ________________________________ These are the lines 378-381 from the config file modparam("dispatcher", "dst_avp", "$avp(DISPATCHER_DST_AVP)") modparam("dispatcher", "grp_avp", "$avp(DISPATCHER_GRP_AVP)") modparam("dispatcher", "cnt_avp", "$avp(DISPATCHER_CNT_AVP)") modparam("dispatcher", "sock_avp", "$avp(DISPATCHER_SOCK_AVP)") Best, Maab Sensitivity: Internal [View Less]

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How to resolve these mod_param errors in Kamailio 5.3.1?
by Muhammad Maab Saleem 13 Jun '20

13 Jun '20

This is the error I get when I run ./kamailio -f /etc/kamailio/kamailio.cfg -E ________________________________ loading modules under config path: /usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/:/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/ 0(28369) ERROR: [core/modparam.c:140]: set_mod_param_regex(): parameter <dst_avp> of type <1> not found in module 0(28369) CRITICAL: [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, … [View More]

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Dispatcher module and hash algorithm
by Luis Rojas G. 13 Jun '20

13 Jun '20

Hello, I noticed a couple of strange behaviors, using this version : kamailio-5.3.3-4.1.x86_64 I am using algorithm "0". Hash over Call-ID. The reason for that is that I am just re-sending the message, using send_udp(), to a destination in a group from file dispatcher.list. Kamailio is completely stateless, it does not need to remember anything. And using a hash over Call-ID I can be sure that any other message related to the same call will be sent to the same destination&… [View More]nbsp; : retransmissions of the INVITE, CANCEL, ACK, BYE, etc. I general, it works, but I detected two problems. I have a group of 5 destinations, where every server is running at 80% of the nominal load. It's a 4 + 1 configuration, for redundancy. I know it's not 100% evenly distributed using hash over Call-ID, but in practice it's almost perfectly even. However, if one node is down, instead of having the four alive running at 100%, I get three running still at 80% capacity, and one is sent 160% of the load (which of course can't process). it seems, all traffic that was supposed to be sent to the failed node is transferred to the same unique destination. That implementation makes my solution for redundancy worthless. Why didn't the implementation use, for instance, a re-hash over the hash? That would allow to redistribute "evenly" over the remaining servers. I tried to solve this by repeating the entries in dispatcher.list, as I noticed that kamailio doesn't check if different lines contain duplicated uris. For instance, if I have two lines with "destination A", and one line with "destination B", like this: 0 sip:server_A 0 sip:server_A 0 sip:server_B I see that "destination A" receives twice the amount of SIP OPTIONS that "destination B" receives. And so, I used this dispatcher.list : 0 sip:server_1 0 sip:server_2 0 sip:server_3 0 sip:server_4 0 sip:server_5 0 sip:server_1 0 sip:server_3 0 sip:server_5 0 sip:server_2 0 sip:server_4 0 sip:server_1 0 sip:server_4 0 sip:server_2 0 sip:server_5 0 sip:server_3 0 sip:server_1 0 sip:server_5 0 sip:server_4 0 sip:server_3 0 sip:server_2 (it's not a random order. It follows a sequence) I thought : "what a genius I am. This way, if any node fails, and kamailio select the "next" to send the traffic to, it will distribute evenly over the rest". It doesn't work. I detected something that looks like a bug. If one entry is repeated at least once, for instance : 0 sip:server_1 0 sip:server_1 0 sip:server_2 0 sip:server_3 0 sip:server_4 0 sip:server_5 if server_1 is down, kamailio still sends some of the INVITEs to it. Not all of them, but many. I can see the SIP OPTIONS been sent to it, and the response "ICMP Destination Unreachable", so Kamailio knows that server_down. However, still sends INVITE requests to that failed node. Look at this trace : You can see all the failed SIP OPTIONS. Still, kamailio sends traffic to that server. It does not send traffic to a failed destination if it's listed only once in dispatcher.list Any ideas? Thanks in advance, Luis -- Luis Rojas Software Architect Sixbell Los Leones 1200 Providencia Santiago, Chile Phone: (+56-2) 22001288 mailto:luis.rojas@sixbell.com http://www.sixbell.com [View Less]

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RPC Command tm.clean
by Marat Gareev 12 Jun '20

12 Jun '20

Hello. I have a philosophical question about tm.clean <https://kamailio.org/docs/modules/5.1.x/modules/tm.html#tm.rpc.clean> command. Unfortunately, I don’t understand what this command is for, because in my view, expired transactions are automatically deleted. This is all I could find: pc command tm.clean to clean long time expired transactions (for troubleshooting purposes) [1] <https://www.kamailio.org/wiki/features/new-in-5.1.x> So, can you give an example when a … [View More]

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Queries regarding Load balancing setup
by Abdul Rasheed 12 Jun '20

12 Jun '20

Hi , I am a newbie who started working on Kamailio only few days ago. I am trying to setup kamailio in such a way that kamailio acts as a load balancer for 2 or more asterisk systems. i have following queries. 1. As i understand , all the SIP registration will be done with Kamailio and the Media of the calls will be routed to Asterisks in a round robin fashion. In such a case , is it required that all the extension numbers to be used are to be configured in all the asterisk systems included … [View More]

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Prometheus module stat
by David Villasmil 12 Jun '20

12 Jun '20

Hello all, Anyone here uses the prometheus module? There’s a core stat that includes a hyphen (“-“) and Prometheus doesn’t support that, only underscores are allowed kamailio_usrloc_location-contacts I’m thinking maybe I can create a custom counter and set it to that stat but with the correct name, is that possible? -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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facing Call Connection and Disconnection issue in Kamailio SIP Server
by Rajkumar Samikkannu 12 Jun '20

12 Jun '20

Hi There, This is Raj. We have successfully installed Kamailio SIP Server by following this documentation link https://kamailio.org/docs/tutorials/5.3.x/kamailio-install-guide-git/#insta… We created some users and those users are in the Subscription Table. We registered(logged in) some users on our client side. We could get Acknowledgement while we registered(logged in). We couldn't get any acknowledgement while we tried to make a call or send a message. We got a "Request Timeout" Error in … [View More]

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Which mode to capture "written to network" traffic with homer and topos
by Karsten Horsmann 12 Jun '20

12 Jun '20

Hi Mailinglist, some of my kamailio-version is 5.3.4 with topos. During the last discussion about sip_trace_mode("t") i found new behaivor. With this tracingmode i see the "internal" before topos writing. Here my questions: Can i change it to "whats on the wire" (topos rewritten mode)? I didnt checked all modes. And if so, can i switch that on-the-fly with $var for example? For topos debugging its fine with sip_trace_mode("t") - not for "hey carrier, you made some mistakes". #!ifdef … [View More]WITH_HOMER #Siptrace modparam("siptrace", "force_send_sock", "MY_LOCAL_SENDSOCK") # local interface from where to send the duplicated traffic modparam("siptrace", "duplicate_uri", "MY_HOMER_CAPTURE") modparam("siptrace", "hep_mode_on", 1) # enable/disable Homer encapsulate mode modparam("siptrace", "hep_version", 3) # HEPv3 the timestamp and capture agent ID will be included in the HEP header modparam("siptrace", "trace_to_database", 0) # trace to database modparam("siptrace", "trace_on", 1) # enable/disable trace modparam("siptrace", "trace_mode", 0) # default 0, if 1 then you dont need call siptrace flag or sip_trace() modparam("siptrace", "hep_capture_id", 8) # capture agent id Limitation: 32-bit for HEPv3. #!endif #!ifdef WITH_HOMER # Set the tracing mode: message, transaction or dialog. Need 5.3.x sip_trace_mode("t"); #!endif *** captured on the topos kamailio: 2020-06-11 08:35:11 +0200 : 172.24.32.28:5060 -> 172.24.52.3:5060 INVITE sip:+49XXXXXXXXXXXX@172.24.52.3:5060 SIP/2.0 Record-Route: <sip:172.24.32.28;r2=on;lr=on;ftag=1ea69f36;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAO3VzZXI9cGhvbmU-;vst=AAAAAAAAAAAAAAAAAAAAAAAAAABLXVQATFVCVUheUUgcXAIBHnBvcnQ9dGNwO3VzZXI9cGhvbmU-;proxy_media=yes;nat=yes> Record-Route: <sip:100.80.0.20;transport=tcp;r2=on;lr=on;ftag=1ea69f36;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAO3VzZXI9cGhvbmU-;vst=AAAAAAAAAAAAAAAAAAAAAAAAAABLXVQATFVCVUheUUgcXAIBHnBvcnQ9dGNwO3VzZXI9cGhvbmU-;proxy_media=yes;nat=yes> Via: SIP/2.0/UDP 172.24.32.28;branch=z9hG4bK34b.1b6e5fbd9cf7af26bb168d1469096714.0;i=74 Via: SIP/2.0/TCP AA.AAA.AAA.AAA:5060;rport=5060;branch=z9hG4bKeca7a7f789c9f647560353e3cf1cd592.ba7930dc Max-Forwards: 52 To: <sip:+49XXXXXXXXXXXX@some.provider.example> From: <sip:+49YYYYYYYYYYY@some.provider.example>;tag=1ea69f36 Call-ID: 3af87db089119841(a)AA.AAA.AAA.AAA Contact: <sip:TFFRA002@AA.AAA.AAA.AAA:5060;transport=tcp> Supported: histinfo,replaces CSeq: 647169 INVITE Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, REFER, REGISTER, UPDATE P-Asserted-Identity: <sip:+49BBBBBBBBBB@some.provider.example;user=phone> Content-Type: application/sdp Content-Disposition: session Content-Length: 298 X-Group-SBC: 2006 X-Routing-SBC: un2tr-cgn-trunk --------------------------------------- *** captured on the kamailio behind 172.24.52.3 2020-06-11 08:35:11 +0200 : 172.24.32.28:5060 -> 172.24.52.3:5060 INVITE sip:+49XXXXXXXXXXXX@172.24.52.3:5060 SIP/2.0 Via: SIP/2.0/UDP 172.24.32.28;branch=z9hG4bK34b.1b6e5fbd9cf7af26bb168d1469096714.0;i=74 Max-Forwards: 52 To: <sip:+49XXXXXXXXXXXX@some.provider.example> From: <sip:+49YYYYYYYYYYY@some.provider.example>;tag=1ea69f36 Call-ID: 3af87db089119841(a)AA.AAA.AAA.AAA Supported: histinfo,replaces CSeq: 647169 INVITE Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, REFER, REGISTER, UPDATE P-Asserted-Identity: <sip:+49BBBBBBBBBB@some.provider.example;user=phone> Content-Type: application/sdp Content-Disposition: session Content-Length: 298 X-Group-SBC: 2006 X-Routing-SBC: un2tr-cgn-trunk Contact: <sip:btpsh-5ee1105a-1903c-1@172.24.32.28> -- Cheers *Karsten Horsmann* [View Less]

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Re: [SR-Users] [EXT] RE: kamailio 522 S-CSCF failing to start
by Martin W Woscek 12 Jun '20

12 Jun '20

What version of MySQL should we be at for kamailio 5.2.2? Thanks, Martin ________________________________ From: Henning Westerholt <hw(a)skalatan.de> Sent: Monday, June 8, 2020 4:49:54 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Cc: Martin W Woscek <mwoscek(a)mitre.org> Subject: [EXT] RE: kamailio 522 S-CSCF failing to start Hello, the error message below is from the DB startup version check. So do you have sometimes DB errors on this … [View More]virtual machine? Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Martin W Woscek Sent: Monday, June 8, 2020 2:46 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: [SR-Users] kamailio 522 S-CSCF failing to start Hello, There are times that our S-CSCF module just wont start. I’ve tried correlating the issue with some operation such as not de-registering an IMS client (Boghe or imsdroid) prior to shutting down the Kamailio server. But this isn’t always the case. Being that we are running this an a Ubuntu Virtual Machine, simply overwriting the image with a backed up VM is the temporary solution. I’ve dropped and created the db for S (kamdbctl) and rearranged the order of loading the presence module with same result. Status of the S-CSCF daemon on failure to start: # systemctl status kamailio_scscf ● kamailio_scscf.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio_scscf; generated) Active: inactive (dead) Docs: man:systemd-sysv-generator(8) Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jun 08 08:22:55 ubuntu kamailio_scscf[16495]: * already running Jun 08 08:22:55 ubuntu kamailio_scscf[16495]: ...done. Jun 08 08:22:55 ubuntu systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Jun 08 08:24:03 ubuntu systemd[1]: Stopping LSB: Start the Kamailio SIP proxy server... Jun 08 08:24:03 ubuntu kamailio_scscf[16517]: * Stopping Kamailio IMS S-CSCF: kamailio_scscf Jun 08 08:24:03 ubuntu kamailio_scscf[16517]: ...done. Jun 08 08:24:03 ubuntu systemd[1]: Stopped LSB: Start the Kamailio SIP proxy server. root@ubuntu:~/IMS_manual_scripts# systemctl start kamailio_scscf root@ubuntu:~/IMS_manual_scripts# systemctl status kamailio_scscf ● kamailio_scscf.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio_scscf; generated) Active: active (exited) since Mon 2020-06-08 08:27:23 EDT; 58s ago Docs: man:systemd-sysv-generator(8) Process: 16543 ExecStart=/etc/init.d/kamailio_scscf start (code=exited, status=0/SUCCESS) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: <core> [db.c:457]: db_check_table_version(): invalid version 0 for table presentity found, expected 5 (check table structure and ta Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: presence [presence.c:367]: mod_init(): error during table version check Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: <core> [core/sr_module.c:898]: init_mod(): Error while initializing module presence (/usr/local/lib64/kamailio/modules/presence.so) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: WARNING: <core> [core/mem/q_malloc.c:480]: qm_free(): WARNING: free(0) called from cdp_avp: cdp_avp_mod.c: cdp_avp_destroy(226) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jun 08 08:27:23 ubuntu kamailio_scscf[16543]: * already running Jun 08 08:27:23 ubuntu kamailio_scscf[16543]: ...done. Jun 08 08:27:23 ubuntu systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Any insight is appreciated. Thanks, _Martin [View Less]

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How to change contact in SIP INVITE
by Trần Đức Thành 09 Jun '20

09 Jun '20

Hi, Anyone can help Sincerely, ---- TRAN DUC THANH Technician ZLINK TECHNOLOGY CO., Ltd Add: 4th Floor, 14 Phao Dai Lang Str, Dong Da Dist., Hanoi, Vietnam 186 Bis Tran Quang Khai, Dist 1, HCM, Vietnam Tel: (84-24) 7 309 99 68 / (84-28) 7309 99 68 Mobile: (84)988612049 - (84)904706808 - Skype ID: thanhtd07 Email: [ mailto:tranthanh@voip.com.vn | tranthanh(a)voip.com.vn ] ZLINK - VoIP Solution - Software Licensing - Server&Network Device [ http://voip.com.vn/ | VoIP.com.vn ] … [View More]

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Reconnect call
by Gaurav Bmotra 09 Jun '20

09 Jun '20

hi how we can implement , that if call got disconnected due to low network and will reconnect .. pl help me -- *Regards:* Gaurav Kumar

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Does DMQ auto-discover cluster members + how?
by George Diamantopoulos 09 Jun '20

09 Jun '20

Hello all (again), When running dmq.list_nodes on one of my production kamailio servers, I get several entries corresponding to staging instances of kamailio. These staging instances are located on separate broadcast domains (different VLANs), and none of them are configured in the notification_address parameter for the dmq module. How do they end up there, and how do I get rid of them? Thanks! George

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RTPProxy Traversal for all Calls
by Mike B 09 Jun '20

09 Jun '20

All, Hopefully a quick answer on this one. I am looking for away to have all traffic traverse the RTP relay, not only NAT. Thinking there should be an easy way to do this with RTPPROXY but haven't been able to figure it out. Using Kamailio 5.3 with RTPProxy 2.2 Thanks, Mike

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Homer 7 and Kamailio
by Social Boh 09 Jun '20

09 Jun '20

Hello, Does Kamailio works with homer sip capture server 7 and HEPv3? I'm trying to test without success. Regards -- --- I'm SoCIaL, MayBe

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Re: [SR-Users] Issue of calls drop 32s
by Rex Lin (林昱頡) 09 Jun '20

09 Jun '20

Hi Alex, Thanks for your reply. ICE/STUN haven’t been activated by our clients. All transmission traffic pass through a proxy server. Below is the screenshot of a normal call process. [cid:image001.png@01D62AC5.FBF723D0] Best Regards, Rex -----Original Message----- From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Alex Balashov Sent: Thursday, May 14, 2020 12:18 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [… [View More]

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kamailio 522 S-CSCF failing to start
by Martin W Woscek 08 Jun '20

08 Jun '20

Hello, There are times that our S-CSCF module just wont start. I’ve tried correlating the issue with some operation such as not de-registering an IMS client (Boghe or imsdroid) prior to shutting down the Kamailio server. But this isn’t always the case. Being that we are running this an a Ubuntu Virtual Machine, simply overwriting the image with a backed up VM is the temporary solution. I’ve dropped and created the db for S (kamdbctl) and rearranged the order of loading the presence module … [View More]with same result. Status of the S-CSCF daemon on failure to start: # systemctl status kamailio_scscf ● kamailio_scscf.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio_scscf; generated) Active: inactive (dead) Docs: man:systemd-sysv-generator(8) Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jun 08 08:22:55 ubuntu kamailio_scscf[16495]: * already running Jun 08 08:22:55 ubuntu kamailio_scscf[16495]: ...done. Jun 08 08:22:55 ubuntu systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Jun 08 08:24:03 ubuntu systemd[1]: Stopping LSB: Start the Kamailio SIP proxy server... Jun 08 08:24:03 ubuntu kamailio_scscf[16517]: * Stopping Kamailio IMS S-CSCF: kamailio_scscf Jun 08 08:24:03 ubuntu kamailio_scscf[16517]: ...done. Jun 08 08:24:03 ubuntu systemd[1]: Stopped LSB: Start the Kamailio SIP proxy server. root@ubuntu:~/IMS_manual_scripts# systemctl start kamailio_scscf root@ubuntu:~/IMS_manual_scripts# systemctl status kamailio_scscf ● kamailio_scscf.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio_scscf; generated) Active: active (exited) since Mon 2020-06-08 08:27:23 EDT; 58s ago Docs: man:systemd-sysv-generator(8) Process: 16543 ExecStart=/etc/init.d/kamailio_scscf start (code=exited, status=0/SUCCESS) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: <core> [db.c:457]: db_check_table_version(): invalid version 0 for table presentity found, expected 5 (check table structure and ta Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: presence [presence.c:367]: mod_init(): error during table version check Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: <core> [core/sr_module.c:898]: init_mod(): Error while initializing module presence (/usr/local/lib64/kamailio/modules/presence.so) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: WARNING: <core> [core/mem/q_malloc.c:480]: qm_free(): WARNING: free(0) called from cdp_avp: cdp_avp_mod.c: cdp_avp_destroy(226) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jun 08 08:27:23 ubuntu kamailio_scscf[16543]: * already running Jun 08 08:27:23 ubuntu kamailio_scscf[16543]: ...done. Jun 08 08:27:23 ubuntu systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Any insight is appreciated. Thanks, _Martin [View Less]

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RTP port range in kuberenetes for kamailio
by Gaurav Bmotra 08 Jun '20

08 Jun '20

hi i have issue audio issue with kamailio in kubernetse cluster anybody , idea how to fix it in kubernetses thank you -- *Regards:* Gaurav Kumar

3 4

How to change contact in SIP INVITE
by Trần Đức Thành 08 Jun '20

08 Jun '20

Hi everyone, I use kamailio and have some problem, need support. My scenario: - Kamailio ( x.x.x.x ) --> SIP provider (y.y.y.y) When i callout, SIP INVITE have 2 Record Route (1 x.x.x.x , 1 127.0.0.1) and 2 Via (1 x.x.x.x , 1 127.0.0.1), SIP provider reject it. I try to enable topos, SIP INVITE now have 1 Record Route x.x.x.x , 1 Via x.x.x.x. Everything look ok but Contact be masked like <sip:abcdefghijk@x.x.x.x> and SIP Provider reject it. Any one know how to have Contact … [View More]

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Re%3A RPC Command tm.clean&In-Reply-To=<9c051ed0-20b5-cf64-3e9a-cc60111a7e7c%40gmail.com>
by Marat Gareev 06 Jun '20

06 Jun '20

Hi, Thanks for your reply! What do you mean by "freezing operation system"? Can I manually reproduce stalled transactions? I tried to send SIGSTOP to a group of Kamailio processes while the transaction is live. The idea is to freeze processes for a while, then continue to work, so that by this time the transaction deletion time was missed. But after a few seconds, the transaction is still cleared.

1 0

insert new timestamp value to cdr table
by mohsen khashei 05 Jun '20

05 Jun '20

Hi I want to add a new column to acc table.I created a new column in the acc table of type timestamp and named it ring_time. in every call i put the ring time to a $dlg_var like this: $dlg_var(ringtime) = $Ts; then I add a extra column in config like this: modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;ring_time=$dlg_var(ringtime)") but when I try to test it always i get: db_mysql [km_dbase.c:122]: db_mysql_submit_query(): … [View More]

2 1

How to overwrite DisplayName
by Stefano Larosa 05 Jun '20

05 Jun '20

Hi and first of all thanks for your amazing work. I have a kamailio 5.1.1 and i need to change the DisplayName before sending INVITE to another Asterisk server. I need that "Sergio"<+33558700000(a)80.169.227.18> Becomes "Simona"<+33558700000(a)80.169.227.18> I've tried just changing $fn value: $fn="Simona" Or using uac_replace_from("Simona ",""); But INVITE msg is always From: "Sergio"<sip:+33558700000@80.169.227.18;transport=UDP>;tag=… [View More]

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Kamailio under NAT
by Aidar Kamalov 05 Jun '20

05 Jun '20

Hello, I am trying to start kamailio+rtpengine+asterisk in docker with docker-compose (not in network host mode). And I faced behavior that I can't understand. When my phone(fanvil) sends a register to kamailio with his ip in contact - 192.168.88.55 (to host's ip address 192.168.88.55), kamailio 200OK answers with "received=172.18.0.1" param in Via. (172.18.0.1 it is host's docker interface, kamailio's ip is 172.18.0.4) After that my phone unregisters with Expires: 0 and sends a new register … [View More]with 172.18.0.1 in contact. Is that correct? Is there any way to fix this? 2020/06/05 12:05:51.764617 172.18.0.1:46142 -> 172.18.0.4:5060 REGISTER sip:192.168.88.55 SIP/2.0 Via: SIP/2.0/UDP 192.168.88.100:5756;branch=z9hG4bK14305253915361123;rport From: akamvm <sip:1000@192.168.88.55:5060>;tag=1125027099 To: akamvm <sip:1000@192.168.88.55:5060> Call-ID: 2849918516212-27188555511609(a)192.168.88.100 CSeq: 1 REGISTER Contact: <sip:1000@192.168.88.100:5756> Max-Forwards: 70 Expires: 3600 Supported: path User-Agent: Fanvil X3S 2.10.2.6887 0c383e217eb9 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Length: 0 2020/06/05 12:05:51.765716 172.18.0.4:5060 -> 172.18.0.1:46142 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.88.100:5756 ;branch=z9hG4bK14305253915361123;rport=46142;received=172.18.0.1 From: akamvm <sip:1000@192.168.88.55:5060>;tag=1125027099 To: akamvm <sip:1000@192.168.88.55:5060 >;tag=1cbccc00ba5f7d13255bc12726ccacc5.59da Call-ID: 2849918516212-27188555511609(a)192.168.88.100 CSeq: 1 REGISTER Contact: <sip:1000@192.168.88.100:5756>;expires=3600 Server: kamailio (5.2.1 (x86_64/linux)) Content-Length: 0 2020/06/05 12:05:51.779447 172.18.0.1:46142 -> 172.18.0.4:5060 REGISTER sip:192.168.88.55 SIP/2.0 Via: SIP/2.0/UDP 192.168.88.100:5756;branch=z9hG4bK251317807114669601;rport From: akamvm <sip:1000@192.168.88.55:5060>;tag=1125027099 To: akamvm <sip:1000@192.168.88.55:5060> Call-ID: 2849918516212-27188555511609(a)192.168.88.100 CSeq: 2 REGISTER Contact: <sip:1000@192.168.88.100:5756> Max-Forwards: 70 Expires: 0 Supported: path User-Agent: Fanvil X3S 2.10.2.6887 0c383e217eb9 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Length: 0 2020/06/05 12:05:51.780479 172.18.0.4:5060 -> 172.18.0.1:46142 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.88.100:5756 ;branch=z9hG4bK251317807114669601;rport=46142;received=172.18.0.1 From: akamvm <sip:1000@192.168.88.55:5060>;tag=1125027099 To: akamvm <sip:1000@192.168.88.55:5060 >;tag=1cbccc00ba5f7d13255bc12726ccacc5.f669 Call-ID: 2849918516212-27188555511609(a)192.168.88.100 CSeq: 2 REGISTER Server: kamailio (5.2.1 (x86_64/linux)) Content-Length: 0 And new register: 2020/06/05 12:05:51.780287 172.18.0.1:46142 -> 172.18.0.4:5060 REGISTER sip:192.168.88.55 SIP/2.0 Via: SIP/2.0/UDP 192.168.88.100:5756;branch=z9hG4bK2696168232620424454;rport From: akamvm <sip:1000@192.168.88.55:5060>;tag=2649930762 To: akamvm <sip:1000@192.168.88.55:5060> Call-ID: 257831362525663-2610472014851(a)172.18.0.1 CSeq: 1 REGISTER Contact: <sip:1000@172.18.0.1:46142> Max-Forwards: 70 Expires: 3600 Supported: path User-Agent: Fanvil X3S 2.10.2.6887 0c383e217eb9 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Length: 0 2020/06/05 12:05:51.781208 172.18.0.4:5060 -> 172.18.0.1:46142 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.88.100:5756 ;branch=z9hG4bK2696168232620424454;rport=46142;received=172.18.0.1 From: akamvm <sip:1000@192.168.88.55:5060>;tag=2649930762 To: akamvm <sip:1000@192.168.88.55:5060 >;tag=1cbccc00ba5f7d13255bc12726ccacc5.4b3b Call-ID: 257831362525663-2610472014851(a)172.18.0.1 CSeq: 1 REGISTER Contact: <sip:1000@172.18.0.1:46142>;expires=3600 Server: kamailio (5.2.1 (x86_64/linux)) Content-Length: 0 -- Aydar A. Kamalov [View Less]

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Kamailio 5.3.0 and HEPv3 oversized traffic
by Karsten Horsmann 04 Jun '20

04 Jun '20

Hi list, I saw some strange behavior in Kamailio 5.3.0 with HEPv3. My Kamailio serves webrtc and sip tls and captures that to an Homer 7.7 instance. I see only the UAC Parts of the message and not the answers from Kamailio in that scenario. Maybe I reached the max Paket size for udp. Hard to debug. In production I saw everything (but there I used and older Kamailio 5.2.x and Homer 5.5 which can handle fragment udp). Any ideas how to hunt the root cause of the problem? Cheers Karsten

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kamcmd mtree.match
by Volker Denneberg 03 Jun '20

03 Jun '20

Hi, given this mtree kamcmd mtree.list siteId { tname: siteId tprefix: 00492419697994 tvalue: { 00492419697994 2388 } } { tname: siteId tprefix: 492415310752 tvalue: { 492415310752 2388 } } i wonder why this lookup fails: kamcmd mtree.match siteId 49241531075212 0 error: 500 - Invalid Parameters My expectation was that 2nd tprefix should match. I am using modparam("mtree", "db_table", "… [View More]

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Re: [SR-Users] Request documentation of Path Flow of Kamailio
by Daniel Cassar 03 Jun '20

03 Jun '20

Cheers Sent from Outlook Mobile<https://aka.ms/blhgte> ________________________________ From: Henning Westerholt <hw(a)skalatan.de> Sent: Wednesday, June 3, 2020 9:39:36 PM To: Daniel Cassar <Daniel.Cassar.a100127(a)mcast.edu.mt>; Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: RE: Request documentation of Path Flow of Kamailio Hello, you can find many presentations (which includes diagrams etc..) in the archive of past Kamailioworld … [View More]conferences on kamailio.org and also on youtube. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Daniel Cassar <Daniel.Cassar.a100127(a)mcast.edu.mt> Sent: Wednesday, June 3, 2020 9:38 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>; Henning Westerholt <hw(a)skalatan.de> Subject: Re: Request documentation of Path Flow of Kamailio One more thing if you have some diagrams that also be considered as how communication should pass from one client to another from a different network please can you send it to me and thanks. Sent from Outlook Mobile<https://aka.ms/blhgte> ________________________________ From: Daniel Cassar <Daniel.Cassar.a100127(a)mcast.edu.mt<mailto:Daniel.Cassar.a100127@mcast.edu.mt>> Sent: Wednesday, June 3, 2020 9:33:19 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>; Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> Subject: Re: Request documentation of Path Flow of Kamailio Thanks :) Sent from Outlook Mobile<https://aka.ms/blhgte> ________________________________ From: Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> Sent: Wednesday, June 3, 2020 9:20:42 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> Cc: Daniel Cassar <Daniel.Cassar.a100127(a)mcast.edu.mt<mailto:Daniel.Cassar.a100127@mcast.edu.mt>> Subject: RE: Request documentation of Path Flow of Kamailio Hello Daniel, thanks for your interest in Kamailio. Looking forward to see the results of your work, please free to give some feedback if its finished. ?? Being an open source project, there is many documentations available e.g. in our wiki. But most of its is more reference character and probably not something that you can directly use in an academic setting. But there is already some work outside in academia which might be useful for you. Check e.g. this diploma thesis out, which describe the implementation of Kamailio as a telephone system for CERN: https://cds.cern.ch/record/1560053/files/CERN-THESIS-2013-073.pdf?subformat… You can find more works that is using Kamailio e.g. with google scholar: https://scholar.google.com/scholar?hl=en&as_sdt=0%2C5&q=kamailio&btnG= Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of Daniel Cassar Sent: Wednesday, June 3, 2020 8:08 PM To: sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Subject: [SR-Users] Request documentation of Path Flow of Kamailio Dear Sir/Madam, I am a student, doing a dissertation about the performance of VPN throughout VoIP Services whereas I decided to configure Kamailio with Siremis Web User Interface on Linux 18.04LTS. I also am using GNS3 as I am trying to connect the server with 3 different routers as a form of sites with different network. What am trying to do is that from the SIP I would try to call from one site to another and see the performance however I need some informations of the Kamailio Server. I need the path communication of how Kamailio works meaning the traffic, protocols being used and these are multicast or unicast when transfer from one site to another. Is this information available somewhere as I seem not to find it anywhere which is why I'm sending you to find out if you have these documentations that I request. I hope to hear from you soon and thanks. Sincerely, Daniel Cassar [View Less]

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Re: [SR-Users] Request documentation of Path Flow of Kamailio
by Daniel Cassar 03 Jun '20

03 Jun '20

Thanks :) Sent from Outlook Mobile<https://aka.ms/blhgte> ________________________________ From: Henning Westerholt <hw(a)skalatan.de> Sent: Wednesday, June 3, 2020 9:20:42 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Cc: Daniel Cassar <Daniel.Cassar.a100127(a)mcast.edu.mt> Subject: RE: Request documentation of Path Flow of Kamailio Hello Daniel, thanks for your interest in Kamailio. Looking forward to see the results of your work, … [View More]

1 1

Request documentation of Path Flow of Kamailio
by Daniel Cassar 03 Jun '20

03 Jun '20

Dear Sir/Madam, I am a student, doing a dissertation about the performance of VPN throughout VoIP Services whereas I decided to configure Kamailio with Siremis Web User Interface on Linux 18.04LTS. I also am using GNS3 as I am trying to connect the server with 3 different routers as a form of sites with different network. What am trying to do is that from the SIP I would try to call from one site to another and see the performance however I need some informations of the Kamailio Server. I need … [View More]

1 0

Trouble loading multiple javascripts with KEMI
by Benjamín Visón 03 Jun '20

03 Jun '20

I am using the module *app_jsdt *with *KEMI *and I am having trouble loading multiple javascript files. The interpreter only notices the last file that's loaded. What I am trying to do is separate my logic so that everything is not inside the kamailio.conf.js (or the main file the contains the necessary functions if you will). modparam("app_jsdt", "load", "/etc/kamailio/file1.js") modparam("app_jsdt", "load", "/etc/kamailio/file2.js") cfgengine "jsdt" In the example above, only functions in … [View More]

1 0

Polycom looses registration within cluster scenario.
by Zhan Bazarov 03 Jun '20

03 Jun '20

Hello. We faced to an issue with polycom, namely - polycom phones looses registration within SRV/DNS based cluster scenario.. So we have round-robin cluster of SIP-proxy instances behind the same SRV/A records(each instance has the same weight). The initial registration works fine, but after the polycom sends a SUBSCRIBE request to another one sipProxy instance(because of round-robin scenario) - polycom stops sending re-register by expires which we are providing in 200Ok message... So the … [View More]

3 4

Received ipv6 problem
by Ahmed Marsou 03 Jun '20

03 Jun '20

Hi; I'm using the received parameter to send a MESSAGE to unregsiter the first user that is registered to register the second one. Using: *modparam("usrloc", "received_column"* *modparam("registrar|nathelper", "received_avp",* and when I do a: select received from location; as you can see is null. +----------+ | received | +----------+ | NULL | +----------+ In IPV4 is working fine, but in IPV6 is not. Can you tell me how can I get this value? Im going crazy with that. … [View More]

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Can't work around double SDP rewrite issue with rtpengine and config script SDP manipulation
by George Diamantopoulos 03 Jun '20

03 Jun '20

Hello all, I'm facing one of those cases where I need to edit the body of a SIP message, which is then to be fed to rtpengine for processing. Although I've taken every precaution I've read about on this list and elsewhere, I can't prevent the edited line from appearing twice in the outgoing message. The configuration file used is huge, so I'm going to try to provide a high-level overview here. But first, the things (I think) I know to be requirements, and which I have striven to meet: - … [View More]If SDP is to be edited, then all such processing is to be carried out in such a way in the script, so that msg_apply_changes() is run as many times as needed before rtpengine offer/answer/manage is called. - rtpengine offer/answer/manage is to be called only once per script iteration - msg_apply_changes can only be called in a request route, or in the core reply_route (i.e. *not* in tm-managed on_reply_route[XXX] blocks) In my case, additionally the following are true: - SDP processing (other than the one performed by rtpengine) takes place in one common route for all cases where it needs to happen. These are two at the moment in my scenario: - Early in the WITHINDLG route (of the example config file) - After the sanity checks in the reply_route (of the example config file) - msg_apply changes() is called once, for each script iteration: - right before rtpengine_manage() is called, provided that t_is_request_route() returns true (so that I don't accidentally call it from a branch route or anything) - rtpengine_manage() is called in its own route, which is very similar to the example config file's "NATMANAGE" route. Since NATMANAGE is called in all branch and on_reply_routes, I employ t_is_request_route() here to make sure it won't execute in those cases. - at the end of the "core" reply_route Now regarding the actual config-file-controlled SDP manipulation, it only consists of a single call to replace_body_str(). The purpose is to edit a line in the message body from something like: - a=fmtp:101 0-16 to something along the lines of: - a=fmtp:101 0-15 For replies, this works as expected. For in-dialog requests, however, I end up with both the original and the edited lines: a=fmtp:101 0-16 (the original line) ... other SDP stuff ... a=fmtp:101 0-15 (the edited line) If anyone could point out any misconceptions I have about msg_apply_changes, SDP rewriting from the script and rtp_engine_X() interoperability, I would be more than grateful. Thank you in advance and I apologize for the long read. Best regards, George Diamantopoulos [View Less]

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Meaning of shmem stats and question on dmq dialog sharing behaviour
by George Diamantopoulos 02 Jun '20

02 Jun '20

Hello all, I've recently been experimenting with dmq, and I've noticed something that seemed a little odd. shmem use would often be abnormally high on otherwise idle kamailio nodes (literally, only dispatcher module OPTIONS were exchanged at the time besides KDMQ messages). After disabling the "enable_dmq" parameter for the dialog module, this sort of absurd behaviour ceased. To get a quantified idea of this change's effect, check out the following screenshot from our monitoring system: https:… [View More]

2 2

rtpengine on different server
by Gaurav Bmotra 02 Jun '20

02 Jun '20

hi i m using kamailio 5.3 ans install rtpengine on different server kindly help me what will be the configuration in case of kamailio and rtpengine thank you -- *Regards:* Gaurav Kumar

2 1

No module matching <http_client> found, failed to load module
by keffa＠alcall.net 02 Jun '20

02 Jun '20

Hello @everybody i can't load http_client module, can someone advice ? ERROR: No module matching <http_client> found, failed to load module version: kamailio 5.3.3 ( installed from GIT ) libcurl4-openssl-dev is installed too in Debian 9 thank for your attention

2 1

record_route_preset and set_advertised_address in branch_route
by Володимир Іванець 29 May '20

29 May '20

Hello! I'm having interesting behavior and wanted to ask if someone knows if this is correct or not. Kamailio v5.3.3. is used. Asterisk sends INVITE to Kamailio. Kamailio performs lookup. Some contacts require Record-Route and Via headers adjustments before sending out INVITEs. I use functions *record_route_preset* and *set_advertised_address* in the branch_route. Everything works as expected when branch 0 does not require changes. But when it does, changes are applied to all other branches … [View More]

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command #invite fails in imc module
by Chris Atta 29 May '20

29 May '20

i'm using pjsua to create a private chat and invite a user: chat was created but i'm not able to add a user. Please, help me! #create chat-020 private imc [imc.c:499]: ki_imc_manager(): BODY #create chat-020 private May 29 16:49:48 lab /usr/sbin/kamailio[7366]: DEBUG: imc [imc.c:502]: ki_imc_manager(): found command May 29 16:49:48 lab /usr/sbin/kamailio[7366]: DEBUG: imc [imc_cmd.c:144]: imc_parse_cmd(): command: [create] May 29 16:49:48 lab /usr/sbin/kamailio[7366]: DEBUG: imc [imc_cmd.c:149]… [View More]

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ds_is_from_list with domain
by Abdirahman A. Osman 29 May '20

29 May '20

Hello, I am using freeswitch in kubernetes and IPaddresses are not fixed instead I am using domain names in the dispatcher.list. Also I am using ds_is_from_list to set the destination to either upstream or downstream. The issue I have is that when freeswitch sends an invite message to Kamailio it is sending IP address and dispatcher.list is in domain format. What will be the best way to match these two? Currently calls are failing with 404 No destination Here # Dispatch requests route[… [View More]

2 1

Kamailio 5.3 - Topos errors when it is processing a cancel
by José Seabra 29 May '20

29 May '20

Hello there, I have been noticing TOPOS errors when kamailio is processing a CANCEL: INFO: ESP_LOG: 752f4a957266379479eee74924687bc40ae1509c: REQ M=CANCEL RURI=sip:333@domain:5060 F=sip:111@domain T=sip:333@domain oP=UDP pr=udp rP=UDP cseq=102 ct=<null> ua=<null> SRC:xx.xxx.xx.xx:5060 => RCV:xx.xx.xx.xx:5060 INFO: ESP_LOG: 752f4a957266379479eee74924687bc40ae1509c: [INTERNAL_TO_EXTERNAL_RELAY] Internal to External relay F=sip:111@xx.xx.xx.xx R=sip:333@domain:5060 D=<null> … [View More]

1 0

congestion control priority for priority based dispatching
by Julien Chavanton 28 May '20

28 May '20

Hi Folks, Just found another simple use case of the dispatcher latency stats. Just shraing this idea of a feature I want to contribute. When using dispatcher algorithm 8: “8” - select destination sorted by priority attribute value (serial forking ordered by priority). If the gateway as the attribute, the priority becomes the estimated latency. cc-priority - The dispatcher would automatically prioritize the closest one. - If a gateway is becoming unresponsive it will automatically … [View More]

3 3

Kamailio not using system resolver for hostname resolution (DMQ)?
by George Diamantopoulos 28 May '20

28 May '20

Hello all, I've come across this today when trying to setup DMQ replication: root@sbcpub0-stage-lhe0-cn1:/root# systemctl restart kamailio root@sbcpub0-stage-lhe0-cn1:/root# kamcmd dmq.list_nodes { host: 172.30.154.189 port: 5090 resolved_ip: 172.30.154.189 status: active last_notification: 0 local: 1 } root@sbcpub0-stage-lhe0-cn1:/root# getent hosts dmq-ng.services.domain.com 172.30.154.189 dmq-ng.services.domain.com 172.17.130.13 dmq-ng.… [View More]

5 10

Fwd: Kamailio--Sipcapture--Homer
by Uttam Giri 27 May '20

27 May '20

Hi,i did setup homer server and wanted to send my sipcapture from Kamailio to homer.however in the Kamailio log I am seeing these errors. Could you please guide how I can fix these errors kamailio log: 2(25) ERROR: <core> [core/udp_server.c:594]: udp_send(): sendto(sock, buf: 0x7f390dc5d438, len: 1525, 0, dst: (10.0.1.59:9060), tolen: 16) - err: Message too long (90) 2(25) ERROR: siptrace [../../core/forward.h:228]: msg_send_buffer(): udp_send failed 2(25) ERROR: siptrace [… [View More]

1 0

Login Attempts
by Ahmed Marsou 27 May '20

27 May '20

Hi; I have to control the login attempts, but blocking the user until I unlock him from web site. To do that I have to save the paramter on database. There is any table and column where I can save that parameter? I'm using the htable module to count the Authentification attempts. There's is any module that do that easily? Thank you.

2 4

CRITICAL: <core> [core/data_lump.c:292]: del_lump(): offset exceeds message size (1436 > 1432) aborting...
by Denys Pozniak 27 May '20

27 May '20

Hello! Could you please check the reason of crash below? syslog: kamailio[31738]: CRITICAL: |RGFLAFL63U|<core> [core/data_lump.c:292]: del_lump(): offset exceeds message size (1436 > 1432) aborting... gdb ( uri and IP were modified ): Core was generated by `/usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.'. Program terminated with signal 6, Aborted. #0 0x00007f5d8556d4f5 in raise () from /lib64/libc.so.6 Missing separate debuginfos, use: debuginfo-… [View More]install cyrus-sasl-lib-2.1.23-15.el6_6.2.x86_64 glibc-2.12-1.212.el6.x86_64 keyutils-libs-1.4-5.el6.x86_64 krb5-libs-1.10.3-65.el6.x86_64 libcom_err-1.41.12-24.el6.x86_64 libcurl-7.19.7-53.el6_9.x86_64 libidn-1.18-2.el6.x86_64 libselinux-2.0.94-7.el6.x86_64 libssh2-1.4.2-2.el6_7.1.x86_64 mysql-libs-5.1.73-8.el6_8.x86_64 nspr-4.19.0-1.el6.x86_64 nss-3.36.0-9.el6_10.x86_64 nss-softokn-freebl-3.14.3-23.3.el6_8.x86_64 nss-util-3.36.0-1.el6.x86_64 openldap-2.4.40-16.el6.x86_64 openssl-1.0.1e-58.el6_10.x86_64 zlib-1.2.3-29.el6.x86_64 (gdb) bt #0 0x00007f5d8556d4f5 in raise () from /lib64/libc.so.6 #1 0x00007f5d8556ecd5 in abort () from /lib64/libc.so.6 #2 0x000000000046a4a6 in del_lump (msg=0x7f5d846cd798, offset=1436, len=29, type=HDR_OTHER_T) at core/data_lump.c:293 #3 0x00007f5d7e6fd7f5 in alter_rtcp (msg=0x7f5d846cd798, body=0x7fffc60c8600, oldport=0x7f5d84715500, newport=0x7fffc60c8550) at rtpproxy.c:1229 #4 0x00007f5d7e70ec5d in force_rtp_proxy (msg=0x7f5d846cd798, str1=0x7f5d846af190 "co", str2=0x7fffc60c87f0 "162.221.21.250", offer=1, forcedIP=0) at rtpproxy.c:2669 #5 0x00007f5d7e705f64 in rtpproxy_manage (msg=0x7f5d846cd798, flags=0x7f5d846af190 "co", ip=0x0) at rtpproxy.c:1862 #6 0x00007f5d7e7064c6 in rtpproxy_manage1 (msg=0x7f5d846cd798, flags=0x7f5d846907f0 "\220\361j\204]\177", ip=0x0) at rtpproxy.c:1904 #7 0x0000000000434f19 in do_action (h=0x7fffc60cae70, a=0x7f5d846b16d8, msg=0x7f5d846cd798) at core/action.c:1073 #8 0x0000000000441a94 in run_actions (h=0x7fffc60cae70, a=0x7f5d846b16d8, msg=0x7f5d846cd798) at core/action.c:1564 #9 0x0000000000431663 in do_action (h=0x7fffc60cae70, a=0x7f5d8468ae40, msg=0x7f5d846cd798) at core/action.c:691 #10 0x0000000000441a94 in run_actions (h=0x7fffc60cae70, a=0x7f5d846848a0, msg=0x7f5d846cd798) at core/action.c:1564 #11 0x0000000000431663 in do_action (h=0x7fffc60cae70, a=0x7f5d846abca0, msg=0x7f5d846cd798) at core/action.c:691 #12 0x0000000000441a94 in run_actions (h=0x7fffc60cae70, a=0x7f5d846abca0, msg=0x7f5d846cd798) at core/action.c:1564 #13 0x0000000000434df4 in do_action (h=0x7fffc60cae70, a=0x7f5d846abdf0, msg=0x7f5d846cd798) at core/action.c:1058 #14 0x0000000000441a94 in run_actions (h=0x7fffc60cae70, a=0x7f5d846a45f8, msg=0x7f5d846cd798) at core/action.c:1564 #15 0x0000000000442190 in run_top_route (a=0x7f5d846a45f8, msg=0x7f5d846cd798, c=0x0) at core/action.c:1646 #16 0x0000000000546d87 in receive_msg ( buf=0x29463b0 "INVITE sip:999@18.12.188.108 SIP/2.0\r\nVia: SIP/2.0/TCP 10.0.0.33:49631;branch=z9hG4bK.9XHFYH0zF;rport\r\nFrom: < sip:RGFLAFL63U@proxy.company.com>;tag=DcTh-~6~M\r\nTo: \"Liz - Home Office\" <s"..., len=1432, rcv_info=0x7f5d5e3ae778) at core/receive.c:340 #17 0x000000000064807d in receive_tcp_msg ( tcpbuf=0x7f5d5e3aea58 "INVITE sip:999@18.12.188.108 SIP/2.0\r\nVia: SIP/2.0/TCP 10.0.0.33:49631;branch=z9hG4bK.9XHFYH0zF;rport\r\nFrom: < sip:RGFLAFL63U@proxy.company.com>;tag=DcTh-~6~M\r\nTo: \"Liz - Home Office\" <s"..., len=1432, rcv_info=0x7f5d5e3ae778, con=0x7f5d5e3ae760) at core/tcp_read.c:1448 #18 0x000000000064a0fd in tcp_read_req (con=0x7f5d5e3ae760, bytes_read=0x7fffc60cb8ac, read_flags=0x7fffc60cb8a4) at core/tcp_read.c:1631 #19 0x000000000064dcd3 in handle_io (fm=0x7f5d846ecda0, events=1, idx=-1) at core/tcp_read.c:1862 #20 0x000000000063b7c4 in io_wait_loop_epoll (h=0xacd340, t=2, repeat=0) at core/io_wait.h:1065 #21 0x000000000064f133 in tcp_receive_loop (unix_sock=51) at core/tcp_read.c:1974 #22 0x00000000006318c8 in tcp_init_children () at core/tcp_main.c:4853 #23 0x0000000000425db5 in main_loop () at main.c:1745 #24 0x000000000042c45e in main (argc=13, argv=0x7fffc60cc0e8) at main.c:2696 (gdb) CentOS release 6.10 (Final) version: kamailio 5.2.2 (x86_64/linux) 67f967 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: 67f967 compiled on 11:39:47 Mar 11 2019 with gcc 4.4.7 -- BR, Denys Pozniak [View Less]

2 1

CANCEL relayed to broadcast 255.255.255.255
by Sergiu Pojoga 27 May '20

27 May '20

Hi gents, Got this weird "problem", where CANCEL request is relayed to both the right destination as well as the 255.255.255.255 broadcast address!? I have no such logic in my script, that's for sure. I doesn't do anything harmful (as far as I can tell), but I'm just curious why it's doing that. Any ideas? version: kamailio 5.3.4 *SIP Trace:* U 2020/05/26 08:40:10.941722 PBXIP:5060 -> KAMAILIO_IP:5060 CANCEL sip:2383@DOWNSTREAM_IP:10533;transport=TLS SIP/2.0 Via: SIP/2.0/UDP PBXIP:5060;… [View More]

2 2

Blind call forwarding with kamailio
by Son Vu 27 May '20

27 May '20

Dear all, I follow tutorial to setup blind call forwarding with kamailio but it seems outdate https://kamailio.org/docs/ser-getting-started/SER-GettingStarted.pdf https://www.kamailio.org/wiki/tutorials/mini-howto-admin/call_forwarding?s%… Is anyone setup it with kamailio? Could you please help me some guidance on this? Best regards, Son -- ================== Best regards, Son Vu

1 0

Using variables in remove_hf
by Cinthia Leung 27 May '20

27 May '20

Hi, I'm trying to use this textops function with a variable, ie remove_hf($var(x)). I know the documentation does not explicitly say that variables are allowed, but it works. However, it's throwing the following error: textops [textops.c:3431]: hname_fixup(): using hdr type name <P-Top-Secret> <core> [core/mem/q_malloc.c:487]: qm_free(): WARNING: free(0) called from core: core/action.c: do_action(1132) Is it something you'd highly recommend against doing? Thanks. Cindy

2 1

High load average
by Володимир Іванець 26 May '20

26 May '20

Hello everyone! I'm running Kamailio version 5.3.3 on a CentOS 6. I started noticing that "load average" value increases rapidly with the start of Kamailio: # uptime 17:47:52 up 4 days, 17:47, 3 users, load average: 7.02, 7.01, 6.02 It will start to decrease immediately after Kamailio is stopped. Does anyone know what could cause this and how to troubleshoot it? Thank you!

4 11

RTPEngine call associations
by Alex Balashov 25 May '20

25 May '20

Hi, When using a single RTPEngine, or a single RTPEngine per set, it is possible to make stateless RTPEngine calls from Kamailio, since everything is keyed by Call-ID + tag. In other words, one can set up a call, answer it, restart Kamailio, and still send a 'delete' to RTPEngine successfully because there's no runtime state that needs to persist within Kamailio. How does this work with multiple RTPEngines in a set? There is clearly some runtime state being kept to map (Call-ID, tag) =&… [View More]

2 1

multi-IP Kamailio
by christian (cx labs) 25 May '20

25 May '20

Hi, you guys helped out so quickly with my last question, so here's me again, with another issue. I need Kamailio to be a sip proxy between two other SIP servers. All trunks are static IP only, no Auth. The specialty is that the Kamailio proxy needs to manipulate different fields in the SIP flow, depending on which public IP is used. So, for example: Provider A, IP 1 -----> Kamailio,* IP A* "change From Header" -----> Provider B, IP 2 Provider A, IP 1 -----> Kamailio, *IP B* "change … [View More]

3 2

488 Not Acceptable Here when remove codecs with rtcp-fb
by José Lopes 25 May '20

25 May '20

Hello, I am using KSR.sdpops.keep_codecs_by_name to remove some codecs. When I make a call between two webrtc clients and I keep only VP8 video codec, the browser rejects the call with 488 Not Acceptable Here. I notice the attributes a=rtcp-fb: of codecs removed are kept. I removed that attributes with: KSR.textops.replace_body_atonce("a=rtcp-fb:1.*\n", ""); KSR.textops.replace_body_atonce("a=rtcp-fb:98.*\n", ""); And the call worked. There is another way to remove these attributes related … [View More]

5 10

IPV6 header in IPV4 : ACK for 200 OK never received
by Abdoul Osséni 25 May '20

25 May '20

Hello, I encounter unexpected behavior. The call flow is: UAC --> Kamailio + Rtpengine (full ipv4) --> Asterisk (full IPV4). I configured the uac to use IPV4 but the SIP INVITES include IPV6 in headers and sdp messages. I think this is a bug on the UAC side. Example: 1. UAC INVITE *INVITE* sip:xxx@sip.xxx.com SIP/2.0 Via: SIP/2.0/TLS [2a04:cec0:101a:ebe2:9949:d81c:23a6:ae10]:42580;branch=*z9hG4bK.I8U1RQDbW* ;rport ..... ..... v=0 o=59130c8f7268a_2 2922 1557 IN IP6 2a04:cec0:101a:… [View More]

2 1

Teams 404
by sip user 25 May '20

25 May '20

Hi, when I try to call Kamailio -> Teams.. this one always respònd to me with a 404 Q.850 cause=1 (Unallocated number). Could be this problem from Teams? Has anyone ever encountered this problem? Thanks

1 0

Change ACK
by sip user 25 May '20

25 May '20

Hi, is it possible change ACK? I have connected Asterisk to Kamailio like extension, with auth and secret. I can redirect tha call from kamailio to my Asterisk but always in ACK, in Asterisk, I received s@..... and in TO number@... is it possible send in ACk the dialed number, like in TO?? Thanks

2 2

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