sr-users October 2014

sr-users@lists.kamailio.org

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187 discussions

BYE issue - Kamailio +RTP proxy
by balu 06 Oct '14

06 Oct '14

Dear experts , I am using kamailio with rtp proxy module. I have 2 questions /issues . 1. When caller or callee ends the call the other end call is not disocnnecting . UA is pjsip based and behind NAT router. Present call flow is pjsipUA (LAN_ip)----->Router (Publicip)-------->Kamailio_with_RTP proxy----> ThridParty SIP Server UA local ip : 192.168.2.11 UA public IP : 89.78.92.23 Kamailio Public ip: 94.50.203.32 Third party Sip server : 76.42.89.25 Here When I disconnect call from either side , it is not disconnecting other side . 2. My second requirement is , how can I define port of third party server . for example if have 3 or 4 sip servers with different sip registration ports other tahn 5060 How can I route registration requests coming from UAs to different ports of third party servers. Please bear my ignorance I am new to kamailio .Hope some experts will help me here . Attached kamailio config and SIP trace taken from kamailio server Thank you INVITE sip:18792356789@76.42.89.25:5060 SIP/2.0 Record-Route: <sip:94.50.203.32:8764 ;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes> Via: SIP/2.0/UDP 94.50.203.32:8764 ;branch=z9hG4bKf7c3.3c8b946c31cdd32f930549d06acfbe9a.0 Via: SIP/2.0/UDP 192.168.2.11:58429 ;received=89.78.92.23;rport=58429;branch=z9hG4bKPjUMSliGvGug7LTZQNqrrpmLN7hggWq7p. Max-Forwards: 69 From: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE To: sip:18792356789@76.42.89.25 Contact: <sip:test@192.168.2.11:58429;ob;alias=89.78.92.23~58429~1> Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz CSeq: 12140 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: Siphon PjSip v2.0.0-beta/arm-apple-darwin9 Content-Type: application/sdp Content-Length: 369 P-hint: outbound v=0 o=- 3620097527 3620097527 IN IP4 94.50.203.32 s=pjmedia c=IN IP4 94.50.203.32 t=0 0 a=X-nat:0 m=audio 60822 RTP/AVP 104 18 0 8 96 c=IN IP4 94.50.203.32 a=rtcp:60823 a=sendrecv a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=nortpproxy:yes SIP/2.0 407 Proxy Authentication Required CSeq: 12140 INVITE Via: SIP/2.0/UDP 94.50.203.32:8764 ;branch=z9hG4bKf7c3.3c8b946c31cdd32f930549d06acfbe9a.0 Via: SIP/2.0/UDP 192.168.2.11:58429 ;branch=z9hG4bKPjUMSliGvGug7LTZQNqrrpmLN7hggWq7p. From: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz To: sip:18792356789@76.42.89.25;tag=19093814023234994340686221 Contact: <sip:76.42.89.25:5060;transport=udp> Proxy-Authenticate: DIGEST realm="sip.testcalls.com", nonce="141110871219020226209383237537" Content-Length: 0 Record-Route: <sip:94.50.203.32:8764 ;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes> ACK sip:18792356789@76.42.89.25:5060 SIP/2.0 Via: SIP/2.0/UDP 94.50.203.32:8764 ;branch=z9hG4bKf7c3.3c8b946c31cdd32f930549d06acfbe9a.0 Max-Forwards: 69 From: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE To: sip:18792356789@76.42.89.25;tag=19093814023234994340686221 Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz CSeq: 12140 ACK Content-Length: 0 INVITE sip:18792356789@76.42.89.25:5060 SIP/2.0 Record-Route: <sip:94.50.203.32:8764 ;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.4f21;nat=yes> Via: SIP/2.0/UDP 94.50.203.32:8764 ;branch=z9hG4bK08c3.4e8266c84727e8ceeaa14c37b4b48ef7.0 Via: SIP/2.0/UDP 192.168.2.11:58429 ;received=89.78.92.23;rport=58429;branch=z9hG4bKPj.fUOV5Ds9oW3oGonVCpIwM9tEpoNUacJ Max-Forwards: 69 From: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE To: sip:18792356789@76.42.89.25 Contact: <sip:test@192.168.2.11:58429;ob;alias=89.78.92.23~58429~1> Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz CSeq: 12141 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: Siphon PjSip v2.0.0-beta/arm-apple-darwin9 Proxy-Authorization: Digest username="test", realm="sip.testcalls.com", nonce="141110871219020226209383237537", uri=" sip:18792356789@76.42.89.25:5060", response="e98f243028e20a3d864dc54149db8ab1" Content-Type: application/sdp Content-Length: 369 P-hint: outbound v=0 o=- 3620097527 3620097527 IN IP4 94.50.203.32 s=pjmedia c=IN IP4 94.50.203.32 t=0 0 a=X-nat:0 m=audio 47574 RTP/AVP 104 18 0 8 96 c=IN IP4 94.50.203.32 a=rtcp:47575 a=sendrecv a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=nortpproxy:yes SIP/2.0 183 Session Progress CSeq: 12141 INVITE Via: SIP/2.0/UDP 94.50.203.32:8764 ;branch=z9hG4bK08c3.4e8266c84727e8ceeaa14c37b4b48ef7.0 Via: SIP/2.0/UDP 192.168.2.11:58429 ;branch=z9hG4bKPj.fUOV5Ds9oW3oGonVCpIwM9tEpoNUacJ From: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz To: sip:18792356789@76.42.89.25;tag=19093814023234994340686221 Contact: <sip:76.42.89.25:5060;transport=udp> Content-Type: application/sdp Content-Length: 224 Record-Route: <sip:94.50.203.32:8764 ;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes> v=0 o=VoipSwitch 6220 7220 IN IP4 76.42.89.25 s=VoipSIP i=Audio Session c=IN IP4 76.42.89.25 t=0 0 m=audio 6220 RTP/AVP 8 96 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=sendrecv SIP/2.0 180 Ringing CSeq: 12141 INVITE Via: SIP/2.0/UDP 94.50.203.32:8764 ;branch=z9hG4bK08c3.4e8266c84727e8ceeaa14c37b4b48ef7.0 Via: SIP/2.0/UDP 192.168.2.11:58429 ;branch=z9hG4bKPj.fUOV5Ds9oW3oGonVCpIwM9tEpoNUacJ From: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz To: sip:18792356789@76.42.89.25;tag=19093814023234994340686221 Contact: <sip:76.42.89.25:5060;transport=udp> Content-Type: application/sdp Content-Length: 224 Record-Route: <sip:94.50.203.32:8764 ;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes> v=0 o=VoipSwitch 6220 7220 IN IP4 76.42.89.25 s=VoipSIP i=Audio Session c=IN IP4 76.42.89.25 t=0 0 m=audio 6220 RTP/AVP 8 96 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=sendrecv SIP/2.0 180 Ringing CSeq: 12141 INVITE Via: SIP/2.0/UDP 94.50.203.32:8764 ;branch=z9hG4bK08c3.4e8266c84727e8ceeaa14c37b4b48ef7.0 Via: SIP/2.0/UDP 192.168.2.11:58429 ;branch=z9hG4bKPj.fUOV5Ds9oW3oGonVCpIwM9tEpoNUacJ From: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz To: sip:18792356789@76.42.89.25;tag=19093814023234994340686221 Contact: <sip:76.42.89.25:5060;transport=udp> Content-Type: application/sdp Content-Length: 224 Record-Route: <sip:94.50.203.32:8764 ;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes> v=0 o=VoipSwitch 6220 7220 IN IP4 76.42.89.25 s=VoipSIP i=Audio Session c=IN IP4 76.42.89.25 t=0 0 m=audio 6220 RTP/AVP 8 96 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=sendrecv SIP/2.0 200 OK CSeq: 12141 INVITE Via: SIP/2.0/UDP 94.50.203.32:8764 ;branch=z9hG4bK08c3.4e8266c84727e8ceeaa14c37b4b48ef7.0 Via: SIP/2.0/UDP 192.168.2.11:58429 ;branch=z9hG4bKPj.fUOV5Ds9oW3oGonVCpIwM9tEpoNUacJ From: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz To: sip:18792356789@76.42.89.25;tag=19093814023234994340686221 Contact: <sip:76.42.89.25:5060;transport=udp> Content-Type: application/sdp Content-Length: 224 Record-Route: <sip:94.50.203.32:8764 ;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes> v=0 o=VoipSwitch 6220 7220 IN IP4 76.42.89.25 s=VoipSIP i=Audio Session c=IN IP4 76.42.89.25 t=0 0 m=audio 6220 RTP/AVP 8 96 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=sendrecv ACK sip:76.42.89.25:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 94.50.203.32:8764 ;branch=z9hG4bK08c3.3f67e202ad7510de3a935364e562755c.0 Via: SIP/2.0/UDP 192.168.2.11:58429 ;received=89.78.92.23;rport=58429;branch=z9hG4bKPjpDeX-U.AQN7jSbWJ7Tqg4Sbl0Lby6nkE Max-Forwards: 69 From: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE To: sip:18792356789@76.42.89.25;tag=19093814023234994340686221 Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz CSeq: 12141 ACK Content-Length: 0 BYE sip:test@192.168.2.11:58429;ob;alias=89.78.92.23%7E58429%7E1 SIP/2.0 Route: <sip:94.50.203.32:8764 ;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes> CSeq: 1 BYE Via: SIP/2.0/UDP 76.42.89.25:5060;branch=z9hG4bK190938140255193499457203 From: sip:18792356789@76.42.89.25;tag=19093814023234994340686221 Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz To: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE Content-Length: 0 Max-Forwards: 70 SIP/2.0 200 OK CSeq: 1 BYE Via: SIP/2.0/UDP 76.42.89.25:5060 ;branch=z9hG4bK190938140255193499457203;rport=5060 From: sip:18792356789@76.42.89.25;tag=19093814023234994340686221 Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz To: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE Server: kamailio (4.1.3 (i386/linux)) Content-Length: 0 ----------- config file below ----------- #!KAMAILIO # check_via=no rev_dns=no dns=no #!define WITH_NAT #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### #!ifdef WITH_MYSQL #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!define WITH_DEBUG #!ifdef WITH_DEBUG debug=2 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=2 memlog=2 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ disable_tcp=yes #sreekanth commented no auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" listen=udp:94.50.203.32:8304 /* port to listen to * - can be specified more than once if needed to listen on many ports */ #port=3074 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules/" #!else mpath="/usr/local/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "mangler.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- 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 ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 1) #sreekanth changed above value from 0 to 1 # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(i:42)") #sreekanth changed above $avp(RECEIVED) to $avp(i:42) modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif #added by sreekanth #!define DLG_FLAG 28 #!define CC_FLAG 29 loadmodule "dialog.so" modparam("dialog", "hash_size", 2048) modparam("dialog", "default_timeout", 3600) modparam("dialog", "db_mode", 0) modparam("dialog", "dlg_flag", DLG_FLAG) loadmodule "rtimer.so"; #!ifdef CNXCC_CHANNEL modparam("rtimer", "timer", "name=ta;interval=1;mode=1;") modparam("rtimer", "exec", "timer=ta;route=SHOW_CHANNEL_COUNT") #!endif loadmodule "cnxcc.so" modparam("cnxcc", "dlg_flag", CC_FLAG) modparam("cnxcc", "credit_check_period", 1) #check every 1 second #sreekanth-End ####### 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 { #setflag(DLG_FLAG); # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); if (is_method("INVITE")) { route(RELAY); } # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) t_check_trans(); # authentication #route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { sl_send_reply("100", "Trying"); setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) { t_on_branch("MANAGE_BRANCH"); } } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); 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()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); #t_newtran(); #t_reply("200", "OK"); # exit; } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) set_contact_alias(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage("co"); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } #if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) #{ # rtpproxy_manage(); # return; #} #from here sreekanth commented #if (is_request()) { #rtpproxy_manage("co"); } #if (is_reply()) { #rtpproxy_manage("z50"); } #if (is_request()) { # if (!has_totag()) { # if(t_is_branch_route()) { # if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB))) { # add_rr_param(";nat=yes"); # } # } # } # } #sreekanth till here if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif }

1 0

Simplest config possible [cont]
by White, Phil 06 Oct '14

06 Oct '14

Hi All, I've had an 'interesting' weekend reading RFCs and playing with kamailio. I'm making progress - however, I would be grateful for a little clarification. I have sorted out a very basic stateless redirect proxy (config file below). I have cut it down to the absolute basic minimum - and it mostly appears to do what I would expect it to do. It does allow me to call any registered extension from my SIP softphone. Problem 1: I have a Grandstream Budgetone 200, which will happily receive calls, but will not initiate any. Looking at a ngrep, it is ignoring the redirect, and sends out a flood of UDP INVITEs (also see below for details) Question: Is this the fault of my configuration, or the Grandstream behaving incorrectly? Problem 2: I am having difficulty in working out which module and command I require to perform a function. Could someone please indicate the INVITE route in this file for a stateful proxy? Many thanks, Phil --- ## WARNING - DO NOT USE THIS IN A PRODUCTION SERVER. FOR TESTING & EDUCTION ONLY debug=2 fork=no log_stderror=yes listen=192.168.1.254:5060 ####### Modules Section ######## mpath="/usr/lib/kamailio/modules/" loadmodule "pv.so" loadmodule "sl.so" loadmodule "usrloc" modparam("usrloc", "db_mode", 0) loadmodule "xlog.so" loadmodule "registrar.so" ####### Routing Logic ######## route{ if(method=="REGISTER") { if (!save("location")) { xlog("L_ERR", "**REGISTER: save failed \n"); sl_reply_error(); } else { xlog("L_NOTICE", " REGISTER rcvd $si:$sp \n"); }; }; if(method=="INVITE") { lookup("location"); switch ($retcode) { case -1: case -3: sl_send_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Not Found"); exit; }; sl_send_reply("302","Redirect"); }; if(method=="PUBLISH" || method=="SUBSCRIBE"){ xlog("L_INFO","REJECTION. We do not process $rm requests. \n"); route(No); }; } route[No] { sl_send_reply(503, "Service Unavailable"); } --- U 2014/10/05 16:38:04.925388 192.168.123.16:5060 -> 192.168.123.254:5060 INVITE sip:105@192.168.123.10:58833;rinstance=0d82c786ca88f460;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 192.168.123.16:5060;branch=z9hG4bKaf202530a743a053. From: "01782478349" <sip:111@192.168.123.254 ;user=phone>;tag=905632682e68e9b5. To: <sip:105@192.168.123.10:58833;rinstance=0d82c786ca88f460;transport=UDP>. Contact: <sip:111@192.168.123.16:5060;transport=udp;user=phone>. Supported: replaces, timer, 100rel, path. Session-Expires: 180. Min-SE: 90. Call-ID: 358a9652404b14ad(a)192.168.123.16. CSeq: 63317 INVITE. User-Agent: Grandstream BT200 1.1.6.32. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK. Content-Type: application/sdp. Content-Length: 407. <...SNIP...> U 2014/10/05 16:38:04.925895 192.168.123.254:5060 -> 192.168.123.16:5060 SIP/2.0 302 Redirect. Via: SIP/2.0/UDP 192.168.123.16:5060 ;branch=z9hG4bKaf202530a743a053;rport=5060. From: "01782478349" <sip:111@192.168.123.254 ;user=phone>;tag=905632682e68e9b5. To: <sip:105@192.168.123.10:58833 ;rinstance=0d82c786ca88f460;transport=UDP>;tag=71d9875c03b727cbe0217d2db6ac2a49.422e. Call-ID: 358a9652404b14ad(a)192.168.123.16. CSeq: 63317 INVITE. Contact: <sip:105@192.168.123.10:58833 ;rinstance=0d82c786ca88f460;transport=UDP>. Server: My SIP Server. Content-Length: 0.U 2014/10/05 16:38:04.954405 192.168.123.16:5060 -> 192.168.123.254:5060 ACK sip:105@192.168.123.10:58833;rinstance=0d82c786ca88f460;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 192.168.123.16:5060;branch=z9hG4bKaf202530a743a053. From: "01782478349" <sip:111@192.168.123.254 ;user=phone>;tag=905632682e68e9b5. To: <sip:105@192.168.123.10:58833 ;rinstance=0d82c786ca88f460;transport=UDP>;tag=71d9875c03b727cbe0217d2db6ac2a49.422e. Contact: <sip:111@192.168.123.16:5060;transport=udp;user=phone>. Supported: path. Call-ID: 358a9652404b14ad(a)192.168.123.16. CSeq: 63317 ACK. User-Agent: Grandstream BT200 1.1.6.32. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK. Content-Length: 0. . U 2014/10/05 16:38:04.975398 192.168.123.16:5060 -> 192.168.123.254:5060 INVITE sip:105@192.168.123.10:58833;rinstance=0d82c786ca88f460;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 192.168.123.16:5060;branch=z9hG4bKa55f284fa742ad32. From: "01782478349" <sip:111@192.168.123.254 ;user=phone>;tag=9a25392722276c54. To: <sip:105@192.168.123.10:58833;rinstance=0d82c786ca88f460;transport=UDP>. Contact: <sip:111@192.168.123.16:5060;transport=udp;user=phone>. Supported: replaces, timer, 100rel, path. Session-Expires: 180. Min-SE: 90. Call-ID: f2989990cfa8525c(a)192.168.123.16. CSeq: 1445 INVITE. User-Agent: Grandstream BT200 1.1.6.32. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK. Content-Type: application/sdp. Content-Length: 407. . . . etc...

4 5

how to always use rtpproxy-ng in kamailio
by andrew 05 Oct '14

05 Oct '14

Hi, I have one question about kamailio. rtpproxy-ng is installed with kamailio. When client A initiates one call to client B, SDP in invite/200ok are rewritten, where ice candidates for rtpproxy-ng are added. Sometimes rtp packets are relayed through rtpproxy-ng, but in some cases rtp packets pass through route created by ICE algorithm. How to set kamailio, then rtpproxy-ng is always used? B.R. andrew

2 1

Re: [SR-Users] Adding Dynamic Routing Module management to Siremis
by cpcnetworking 04 Oct '14

04 Oct '14

Anybody have some insight or pointers? Thx

2 1

channel parameter on SDP
by Kelvin Chua 03 Oct '14

03 Oct '14

Hi guys, my provider is being a dick. they do not accept doubango's a=rtpmap:0 PCMU/8000/1 what they wanted was a=rtpmap:0 PCMU/8000 is there a way of changing this on kamailio? Kelvin Chua

2 1

Routing calls to a subscribed softphone
by Will Ferrer 03 Oct '14

03 Oct '14

Hi I was wondering if any one had any advice or examples for me of how to get a call to be routed to a subscribed softphone. We have 2 boxes in our testing deployment, a load balancer / sbc and a call processing box. Calls come in to the sbc, and then are passed to the call processing box. The call is analyzed and the branch uri is rewritten to a destination address when applicable for the call (this is how we handle routing of calls to certain numbers -- we do this utilizing custom code and a custom db). This works just fine when the destination sip uri is phone number (in which case we do lcr) or if the destination goes to a remote address. However when the destination is a subscriber: sip:[subscriber_name]@[our_domain_name], the call is passed back to the sbc, which passes it to the callprocessing box, back and forth until a too many hops error occurs. The subscriber I am trying to send the call too does show up under "kamctl ul show". I feel like there is something basic I must be missing here. Does any one have any advice for me? Thank you very much in advance. All the best. Will Ferrer

2 5

Enable SipTrace
by Bruno Emer 03 Oct '14

03 Oct '14

Hello! I just got some kamailio servers working with websockets using jssip in my clients and now I want to get the SIP Messages to better understand how it flows between my servers. I have enabled the "siptrace" module, and I can now save information on my database, but the problem is that I can't read the SIP Messages there, as it saves some strings in my database. The SIP messages appears like this store in the sip_trace table: \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 I am currently using postgres as my database, and to get this I just have: - Define "WITH_SIP_TRACE" in the beginning of my configuration file: #!KAMAILIO #!define WITH_PGSQL ##!define WITH_DEBUG #!define WITH_SIP_TRACE - loadmodule: #!ifdef WITH_SIP_TRACE loadmodule "siptrace.so" #!endif - Configure SIP_TRACE params: #!ifdef WITH_SIP_TRACE # ---- sip_trace params ---- modparam("siptrace", "db_url", "postgres://kamailio:kamailiorw@localhost /kamailio") modparam("siptrace", "trace_on", 1) modparam("siptrace", "trace_flag", 22) #!endif - Get the messages: request_route { sip_trace(); setflag(22); route(REQINIT); ... Does anyone knows if is there something wrong in this configuration, or what can I do to fix this and have the SIP messages stored in my database? Thanks! *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com <https://www.facebook.com/bruno.emer.5> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> <https://twitter.com/brunoemer_> <https://www.flickr.com/photos/122070309@N03/> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_>

2 2

Re: [SR-Users] Kamailio & queue
by Robert Boisvert 03 Oct '14

03 Oct '14

Only with rtpproxy. On Fri, Oct 3, 2014 at 5:25 AM, Marino Mileti <marino.mileti(a)alice.it> wrote: > mohqueue module is also compatible with rtpproxy-ng or only with rtppoxy? > > Because during startup i've noticed this error: > mohqueue : Unable to load rtpproxy_stream2uac > > > > > >There was a typo in the parameter name. If you use default values, just > remove the modparam. > >Otherwise, I pushed the patches to master and 4.1 branches. So you would > need to upgrade to latest git version. > >Cheers, > >Daniel > >>On 02/10/14 10:15, Marino Mileti wrote: > >>I don't need any IVR so I would like to try this module and following doc > i've setup dbtable name parameters: > >>modparam ("mohqueue", "db_qtable", "mohqueues") > >>modparam ("mohqueue", "db_ctable", "mohcalls") > >>but when start Kamailio says: > >>ERROR: <core> [modparam.c:156]: set_mod_param_regex(): > set_mod_param_regex: parameter <db_qtable> of type <1> not found in module > <mohqueue> > >>What's wrong? > > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > --- > Questa e-mail è priva di virus e malware perché è attiva la protezione > avast! Antivirus. > http://www.avast.com > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >

1 0

[OT] About next Kamailio World, May 27-29, 2014, in Berlin
by Daniel-Constantin Mierla 03 Oct '14

03 Oct '14

Hello, we have preliminarily booked May 27-29, 2014 for the next Kamailio World event, hosted again by FhG Fokus Institute in Berlin. It was really hard to find a time frame during April and May 2015, because of different dates for Catholic and Orthodox Easters, 1st of May as well as quite some many public holidays in Germany. I am writing to see if anyone is aware of any major blocker for that time frame. Write me directly and we will see if anything can be done, selected the best options based on constraints. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 1

Kamailio & queue
by Marino Mileti 03 Oct '14

03 Oct '14

I don't need any IVR so I would like to try this module and following doc i've setup dbtable name parameters: modparam ("mohqueue", "db_qtable", "mohqueues") modparam ("mohqueue", "db_ctable", "mohcalls") but when start Kamailio says: ERROR: <core> [modparam.c:156]: set_mod_param_regex(): set_mod_param_regex: parameter <db_qtable> of type <1> not found in module <mohqueue> What's wrong? ----Messaggio originale---- Da: miconda(a)gmail.com Data: 2-ott-2014 9.44 A: "Kamailio (SER) - Users Mailing List"<sr-users(a)lists.sip-router.org>, "Marino Mileti"<marino.mileti(a)alice.it> Ogg: Re: [SR-Users] Kamailio & queue Thats true if you need to answer and do some IVR, otherwise you can hold the INVITE or other request (until it is cancelled or times out). Specific on a call queue, may worth checking the module: - http://kamailio.org/docs/modules/stable/modules/mohqueue.html Holding the invite in config file is possible with: - t_suspend() and then the (index, label) identification of transaction -- see tmx module - add (index,label) to mqueue - when the routing of invite has to be resumed, fetch from the queue and do t_continue() Cheers, Daniel

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sr-users October 2014