sr-users August 2015

sr-users@lists.kamailio.org

79 participants
123 discussions

by Loic Chabert

Hello everyone, I'm trying to reproduce BLF's asterisk fonctionnality with kamailio. With asterisk, on register, NOTIFY message has been sent to all suscribed users. So LED became green. How can i reproduce this function ? I have try to execute a "pua_set_publish()" on register route, but it's not work. My led stay "black", and no notify message has been sent. I have loaded presence, presence_xml, presence_mwi, presence_dialoginfo, pua, pua_dialoginfo, pua_usrloc. Any other module needed ? Thanks, Regards.

8 years, 8 months

webrtc not working in different networks

by Achintha

Hi, I have configured kamailio 4.3.1 with mysql ,rtpproxy,websocket modules , then I tried to call through the two extensions using x-lite phone and get the capture. Call is connected. In that test rtp was going through the rtpproxy server, Then I configured jssip to test webrtc in this setup, call is connected I can communicate through the extensions both video and voice only in LAN. IN different networks call is connected but, voice is not hear (rtp not working). Kindly help me to configure this web rtc with kamailio. -- Best Regards.. Achintha

8 years, 8 months

dmq_usrloc

by Kelvin Chua

is there any function to trigger replication? i enabled the module and dmq is working for my htables but usrloc is not replicating. "kamctl ul show" does not show anything i used ngrep to trace the KDMQ packets, i can see the frequent exchanges but nothing related to usrloc. am i doing something wrong? Kelvin Chua

8 years, 8 months

multiple proxies

by Bruce Lefko

If I want to have multiple kamailio proxies in front of multiple media servers, can I balance them using plain old DNS behind a domain name? If I wanted a specific proxy to be notified about an entire SIP dialog I could set the record route to use the public IP of the proxy that received the initial INVITE. Could I leverage something like route53 tcp healthchecks to know if I should be failing over to another proxy?

8 years, 8 months

multiple proxies

by Bruce Lefko

8 years, 8 months

kamailio presence server using db_mongodb as database

by José Seabra

Hello all, I would like to test my presence server using db_mongodb as database server instead of using mysql, but i don't find any documentation explaining how i can configure mongodb database(data structure) for presence or even for the other kamailio modules. Can someone guide me in this setup? Best Regards -- Cumprimentos José Seabra

8 years, 8 months

Confusion about various TCP expiry and lifetime parameters

by M S

Hi, We have a bit of confusion regarding tcp_connecton_timeout core parameter. The documentation says, -- Lifetime in seconds for TCP sessions. TCP sessions which are inactive for longer than *tcp_connection_lifetime* will be closed by Kamailio. -- However we observe a strange behaviour. 1. The connection is NOT closed by Kamailio unless we additionally set "close_expired_tcp" parameter in usrloc module, 2. Secondly, if we set this parameter to a smaller value say 70 seconds while sip register expiry is 600 seconds (and close_expired_tcp parameter enabled in usrloc module), the connection still remains active (tested after keeping it idle for 120 seconds then sending a sip request on this connection, we expected the request to fail but it does not fails). 3. if we set this parameter to greater then sip register value, e.g. 610 seconds and set close_expired_tcp parameter in usrloc, then we observe disconnect after about 8-10 minutes. Whereas we expect it to continue since user is re-registering every 600 seconds. Can you guys explain why this is happening? What keeps a tcp connection active or makes it inactive? We are using websockets (which use TCP at lower layer) and we observe there is no websocket frame sent or received over the tcp connection, yet it remain active after tcp_connection_lifetime in case 2 above and becomes inactive in case 3. We are using Kamailio v4.3.1 SVN Rev. 4717b5 on Debian Wheezy 32bit OS. Please suggest. Thank you.

8 years, 8 months

IMS Kamailio "477 Unfortunately error on sending to next hop occurred" code sent by the PCSCF

by Victor Huertas Garcia

Hello everybody, I have tried to perform a call between alice and bob (both registered in the same SCSCF and from the same PCSCF). The INVITE goes all the way as it is supposed: Alice-->PCSCF-->SBC-->SCSCF-->ICSCF-->SCSCF-->SBC-->PCSCF. But when it reaches PCSCF on Network to User direction, this SIP proxy answers with a "477 Unfortunately error on sending to next hop occurred" code just after an exchange with the RTP Proxy (Offer-Answer). At kamailio.cfg (which it is not change from the default config) this exchange is done under Route[RTPRPROXY] in the function: rtpengine_manage("trust-address replace-origin replace-session-connection ICE=remove"); Does anybody know why is complaining (see below) about the server on localhost while it is not configured so? Here you can find the logs from PCSCF: Aug 23 15:10:06 ims rtpengine[556]: [3D504184-55D9D43E00072478-7E0DE700] Received command 'offer' from 192.168.0.108:57522Aug 23 15:10:06 ims rtpengine[556]: [3D504184-55D9D43E00072478-7E0DE700] Creating new callAug 23 15:10:06 ims rtpengine[556]: [3D504184-55D9D43E00072478-7E0DE700] Replying to 'offer' from 192.168.0.108:57522Aug 23 15:10:06 ims /usr/sbin/kamailio[3313]: ERROR: *** cfgtrace:request_route=[RTPPROXY] c=[/etc/kamailio/pcscf/kamailio.cfg] l=813 a=2 n=returnAug 23 15:10:06 ims /usr/sbin/kamailio[3313]: ERROR: *** cfgtrace:request_route=[Term_Initial] c=[/etc/kamailio/pcscf/kamailio.cfg] l=1196 a=24 n=t_relayAug 23 15:10:06 ims /usr/sbin/kamailio[3313]: ERROR: <core> [udp_server.c:550]: udp_send(): sendto(sock,0x7f480e27ed70,1358,0,192.168.0.105:49284,16): Invalid argument(22)Aug 23 15:10:06 ims /usr/sbin/kamailio[3313]: CRITICAL: <core> [udp_server.c:555]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Here you can find the configuration from RTP proxy (/etc/default/ngcp-rtpengine-daemon) and you can see that I am not listening the localhost port at "INTERFACES": RUN_RTPENGINE=yes LISTEN_TCP=25060LISTEN_UDP=12222LISTEN_NG=22222LISTEN_CLI=9900INTERFACES="192.168.0.108"TIMEOUT=60SILENT_TIMEOUT=3600PIDFILE=/var/run/ngcp-rtpengine-daemon.pidFORK=yes# TOS=184TABLE=0# NO_FALLBACK=yes# PORT_MIN=30000# PORT_MAX=50000# REDIS=127.0.0.1:6379# REDIS_DB=1# B2B_URL=http://127.0.0.1:8090/# LOG_LEVEL=6# LOG_FACILITY=daemon# LOG_FACILITY_CDR=daemon# LOG_FACILITY_RTCP=daemon# NUM_THREADS=5# DELETE_DELAY=30# GRAPHITE=9006# GRAPHITE_INTERVAL=60# GRAPHITE_PREFIX=myownprefix# MAX_SESSIONS=5000 And finally the pcscf.cfg: # SIP / UDPlisten=udp:eth0:4060#listen=udp:127.0.0.1:4070listen=udp:127.0.0.1:4060# SIP / TCP (Monitoring)#listen=tcp:127.0.0.1:4060# SIP / TCP/TLS#listen=tls:109.239.57.200:5061# SIP / Websocket #!define MY_WS_PORT 80 listen=tcp:eth0:MY_WS_PORT # SIP / Websocket/TLS#!define MY_WSS_PORT 443#listen=tls:109.239.57.200:MY_WSS_PORT #alias=pcscf-1.imscore.orgalias=pcscf.home-domain.net#alias=proxy.imscore.org#alias=tls:"wss-proxy.imscore.org":443 # Port, where we listen to Traffic#!define PORT 4060 # NUEVO Workshop!#!define SBCPORT 5080#!define PCSCF_URL "sip:pcscf.home-domain.net"#!define TCP_PROCESSES 32 #!subst "/NETWORKNAME/home-domain.net/"#!subst "/HOSTNAME/pcscf.home-domain.net/"#!define HOSTNAME_IP pcscf.home-domain.net#!define HOSTNAME_ESC "pcscf\.home-domain\.net" # SIP-Address of capturing node, if not set, capturing is disabled.##!define CAPTURE_NODE "sip:10.1.8.55" # Allowed IPs for XML-RPC-Queries#!define XMLRPC_WHITELIST_1 "127.0.0.1"##!define XMLRPC_WHITELIST_2 "127.0.0.1"##!define XMLRPC_WHITELIST_3 "127.0.0.1" # Databases:#!define DB_URL "mysql://pcscf:heslo@127.0.0.1/pcscf"##!define DB_URL2 "con2=>mysql://pcscf:heslo@127.0.0.1/pcscf"##!define DB_URL "con1=>mysql://pcscf:heslo@192.168.5.1/pcscf"##!define DB_URL2 "con2=>mysql://pcscf:heslo@10.1.27.217/pcscf" #! Optional: Server-URL for Websocket-Requests##!define WEBSOCKET_WEBSERVER "phone.imscore.org" # NUEVO Workshop!#!subst "/ICID_VALUE_PREFIX/P-CSCFabcd/"#!subst "/ICID_GEN_ADDR/127.0.0.1/" # IP-Adress(es) of the RTP-Proxy##!define RTPPROXY_ADDRESS "udp:10.1.2.186:22222 udp:10.1.27.217:22222"#!define RTPPROXY_ADDRESS "udp:192.168.0.108:22222"## Several features can be enabled using '#!define WITH_FEATURE' directives:## *** To run in debug mode:# - define WITH_DEBUG## *** To enable nat traversal execute:# - define WITH_NAT# - define the connection to the RTP-Proxy: RTPPROXY_ADDRESS## *** To force alls calls through the RTP-Proxy# - this will automagically enable NAT-Traversal# - define FORCE_RTPRELAY## *** To enable IPv4/IPv6 Translation (RTPProxy)# - this will automagically enable NAT-Traversal# - define WITH_RTPIPV4## *** To enable TCP support execute:# - define WITH_TCP## *** To enable TLS support execute:# - adjust CFGDIR/tls.cfg as needed# - define WITH_TLS# - this will automagically enable TCP## *** To enable XMLRPC support execute:# - define WITH_XMLRPC# - this will automagically enable TCP## *** To enable anti-flood detection execute:# - adjust pike and htable=>ipban settings as needed (default is# block if more than 16 requests in 2 seconds and ban for 300 seconds)# - define WITH_ANTIFLOOD## *** To enable the Rx-Interface:# - Configure Rx-Diameter-Interface in pcscf.xml# - define WITH_RX## *** To enable a Homer SIP-Capter-Node:# - define CAPTURE_NODE with a proper address## *** To enable support for the SEMS-SBC# - define WITH_SBC# - configure dispatcher-list with a set of SBC's## *** To enable support for Websocket# - define WITH_WEBSOCKET# - this will automagically enable TCP## Enabled Features for this host:#!define WITH_DEBUG#!define WITH_NAT#!define WITH_NATPING#!define FORCE_RTPRELAY##!define WITH_TLS#!define WITH_XMLRPC#!define WITH_ANTIFLOOD##!define WITH_RX##!define WITH_RX_REG##!define WITH_RX_CALL##!define WITH_TCP##!define WITH_SBC##!define WITH_RTPIPV4#!define WITH_SBC_CALL#!define WITH_REGINFO#!define WITH_WEBSOCKET#!define WITH_IMS_HDR_CACHE Thanks a lot for your attention

8 years, 8 months

Kamailio dead after INVITE from SIPDROID

by Kowalski Piotr 3 - Hurt

Hello Community. We have openIMS nodes in our lab. We have Kamailio 4.0 based solution. I registered SIPDROID client (iptel) on Android and JITSI on WINDOWS. When I call from SIPDROID to JITSI following happens: SIP DROID - iptel - client - registration ok, but INVITE attempt towards JITSI (and other clients) does what's below: HSS (every 30 seconds): [root@hss1 kowalp00]# 2015-08-25 12:16:55,468 WARN de.fhg.fokus.diameter.DiameterPeer.transport.Communicator - run Read 3021 before Exception. 2015-08-25 12:16:56,980 ERROR de.fhg.fokus.diameter.DiameterPeer.peer.StateMachine - I_Snd_Conn_Req StateMachine: Peer scscf1.hss.ims.test.pl:3868 not responding to connection attempt 2015-08-25 12:17:26,985 ERROR de.fhg.fokus.diameter.DiameterPeer.peer.StateMachine - I_Snd_Conn_Req StateMachine: Peer scscf1.hss.ims.test.pl:3868 not responding to connection attempt SCSCF: Aug 25 12:24:18 scscf1 /usr/sbin/kamailio[8879]: INFO: <core> [main.c:854]: INFO: signal 15 received Aug 25 12:24:18 scscf1 /usr/sbin/kamailio[8865]: INFO: <core> [main.c:854]: INFO: signal 15 received Aug 25 12:24:18 scscf1 /usr/sbin/kamailio[8803]: INFO: snmpstats [snmpstats.c:410]: The SNMPStats module got the kill signal Aug 25 12:24:18 scscf1 /usr/sbin/kamailio[8803]: INFO: snmpstats [snmpstats.c:414]: Shutting down the AgentX Sub-Agent! Aug 25 12:24:18 scscf1 /usr/sbin/kamailio[8803]: ERROR: ctl [ctl.c:379]: ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Operation not permitted (1) Aug 25 12:25:18 scscf1 /usr/sbin/kamailio[8803]: : <core> [main.c:697]: BUG: shutdown timeout triggered, dying... Kamailio becomes dead and needs restart. Clients become unregistered. A call from JITSI to SIPDROID it's fine. Any ideas? Piotr Kowalski Starszy Architekt Techniczny Tel. +48 22 699 50 93 Kom. +48 519 123 289 Orange Orange Polska Hurt Badania i Rozwój Sieci i Platform Usługowych Wydział Usług Konwergentnych Obrzeżna 7 /p. 309, 02-691 Warszawa Fax. +48 22 857 99 86 <http://www.hurt-orange.pl/> www.hurt-orange.pl - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Czy musisz drukować tę wiadomość? Pomyśl o środowisku. Korzystaj z e-faktury. Treść tej wiadomości jest własnością Orange Polska i zawiera informacje stanowiące tajemnicę przedsiębiorstwa Orange Polska. Jeżeli nie jesteście Państwo jej adresatem, bądź otrzymaliście ją przez pomyłkę, prosimy o powiadomienie o tym nadawcy oraz trwałe jej usunięcie. Orange Polska Spółka Akcyjna z siedzibą i adresem w Warszawie (02-326) przy Al. Jerozolimskich 160, wpisana do Rejestru Przedsiębiorców prowadzonego przez Sąd Rejonowy dla m.st. Warszawy XII Wydział Gospodarczy Krajowego Rejestru Sądowego pod numerem 0000010681; REGON 012100784, NIP 526-02-50-995; z pokrytym w całości kapitałem zakładowym wynoszącym 3.937.072.437 złotych.

8 years, 8 months

Rewrite Invite w/o sdp

by Serge Yuriev

Hello, Our PBX (Cisco CallManager) uses "late offer" sending invites w/o sdp. After receiving provisional reply 18x it answers prack with sdp I'm trying to realize following scenario: 1 Pbx (invite no sdp) 2 kamailio hold and saves invite 3 send 183 to Pbx 4 get prack with sdp 5 construct new invite from saved one and sdp from prack 6 send it to calee Is this possible? How to save invite for future use? -- Serge via mobile

8 years, 8 months

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sr-users August 2015