sr-users October 2010

sr-users@lists.kamailio.org

77 participants
158 discussions

Best ASR with Kamailio
by Yury Naumovich 13 Oct '10

13 Oct '10

Good day. How with Kamailio improve ASR to ~100%, if we have 3-4 wholesale Voip providers? Each can guaranty only ASR 60-70%. Best regards, Yury

2 1

rtp proxy - where to invoke?
by Anders 13 Oct '10

13 Oct '10

Hi, I'm trying to get the RTP proxy to work with Kamailio (1.5), but right now I am getting this error here: Oct 13 16:54:44 vn1031 /usr/local/sbin/kamailio[13622]: ERROR:nathelper:force_rtp_proxy: incorrect port 0 in reply from rtp proxy There is probably something wrong in my cfg - for the use of force_rtp_proxy that part of my config is here below. See anything odd that could lead to the error above? route { ...... route(4); ...... } route[4]{ force_rport(); if (… [View More]

2 1

core:parse error
by Anders 13 Oct '10

13 Oct '10

Hi, I'm getting a bunch of these sequences in my syslog - and have absolutely no idea where to look for the problem. Any pointers, please? Oct 13 08:12:34 vn1031 /usr/local/sbin/kamailio[2821]: ERROR:core:parse_cseq: expecting CSeq EoL Oct 13 08:12:34 vn1031 /usr/local/sbin/kamailio[2821]: ERROR:core:parse_cseq: bad cseq Oct 13 08:12:34 vn1031 /usr/local/sbin/kamailio[2821]: ERROR:core:get_hdr_field: bad cseq Oct 13 08:12:34 vn1031 /usr/local/sbin/kamailio[2821]: ERROR:maxfwd:… [View More]

2 1

kamailio-3.1.0+lenny3 + siptrace
by MÉSZÁROS Mihály 13 Oct '10

13 Oct '10

Hello all! I am getting the following problem with 3.1.0 kamailio: ERROR: <core> [sr_module.c:875]: init_mod(): Error while initializing module siptrace (/usr/lib/kamailio/modules_k/siptrace.so) config: loadmodule "siptrace.so" # ----- siptrace params ----- modparam("siptrace", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("siptrace", "table", "sip_trace") modparam("siptrace", "trace_flag", 22) modparam("siptrace", "traced_user_avp", "$avp(s:user)") modparam("… [View More]

3 15

Kamailio Scalability
by michel freiha 13 Oct '10

13 Oct '10

*Dear All, I'm planning to use Kamailio in a high network traffic environment... I'm sure that it can handle several thousand of calls when using SIP over UDP but I have a doubt regarding TLS/TCPusage with Kamailio...I saw on the website that the latest version fixed that BUG that I had bad experience with and Kamailio is ready now for using TLS over TCP with no problem... My question is, how much TLS/TCP calls Kamailio can handle and does anyone had an experience with it? Regards*

1 0

mtree memory comsumption question
by anthony thomas 13 Oct '10

13 Oct '10

Hello list, I have been testing the new mtree module but I am a little bit confused about the results. On my tests, I randomly generated 500K, 1M and 2M entries in a table in several trees. The size of the db (on disk) ranges from 120 megs to 270 megs and when Kamailio loads it in memory, it jumps from 1.0Gb to 2.8Gb. Is there any way to pre-calculate the total amount of memory that a tree will need? Thanks, A

2 1

Need some help with showing one of my ITSPs that they have a problem.
by Geoffrey Mina 13 Oct '10

13 Oct '10

Hello, I am currently going through an Inter-Op test with a new carrier and there have been some issues. I have my Kamailio server integrated with many tier-1 carriers, so I'm pretty sure I'm doing everything OK. The problem is in the BYE coming from the ITSP, the URI isn't valid, but I am not sure based on RFC3261 exactly why it isn't correct. If anyone can take a quick glance, I would really appreciate it. Thanks! >>>>>>>>>>>>>>>>>>&… [View More]gt;>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>> OUT >>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> INVITE sip:16782247008@67.150.160.10:5060 SIP/2.0 Record-Route: <sip:X.X.X.90;lr;ftag=as4c62e0b9> Via: SIP/2.0/UDP X.X.X.90;branch=z9hG4bK2aa.c3961912.0 Via: SIP/2.0/UDP X.X.X.238:5060;received=X.X.X.238;branch=z9hG4bK10281a99;rport=5060 From: ""2026097878"" <sip:2026097878@mydomain.com<sip%3A2026097878(a)mydomain.com> >;tag=as4c62e0b9 To: <sip:+16782247008@X.X.X.90:5060> Contact: <sip:2026097878@X.X.X.238> Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com CSeq: 102 INVITE User-Agent: G-Tel v1.0 Max-Forwards: 69 Date: Mon, 11 Oct 2010 17:50:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces P-Charge-Info: ""4019921313"" <sip:4019921313@connectfirst.com<sip%3A4019921313(a)connectfirst.com> > Content-Type: application/sdp Content-Length: 244 v=0 o=root 10137 10137 IN IP4 X.X.X.238 s=session c=IN IP4 X.X.X.238 t=0 0 m=audio 11748 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<< IN <<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< SIP/2.0 100 Trying Via: SIP/2.0/UDP X.X.X.90;branch=z9hG4bK2aa.c3961912.0 Via: SIP/2.0/UDP X.X.X.238:5060;received=X.X.X.238;branch=z9hG4bK10281a99;rport=5060 Record-Route: <sip:X.X.X.90;lr;ftag=as4c62e0b9> From: ""2026097878"" <sip:2026097878@mydomain.com<sip%3A2026097878(a)mydomain.com> >;tag=as4c62e0b9 To: <sip:+16782247008@X.X.X.90:5060> Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com CSeq: 102 INVITE Contact: <sip:16782247008@67.150.160.10 <sip%3A16782247008(a)67.150.160.10>> Content-Length: 0 <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<< IN <<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< SIP/2.0 183 Session Progress Via: SIP/2.0/UDP X.X.X.90;branch=z9hG4bK2aa.c3961912.0 Via: SIP/2.0/UDP X.X.X.238:5060;received=X.X.X.238;branch=z9hG4bK10281a99;rport=5060 Record-Route: <sip:X.X.X.90;lr;ftag=as4c62e0b9> From: ""2026097878"" <sip:2026097878@mydomain.com<sip%3A2026097878(a)mydomain.com> >;tag=as4c62e0b9 To: <sip:+16782247008@X.X.X.90:5060>;tag=H3jDp3KZmHNja Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com CSeq: 102 INVITE Contact: <sip:16782247008@67.150.160.10 <sip%3A16782247008(a)67.150.160.10>> Content-Type: application/sdp Content-Length: 109 v=0 o=ATSI-B1 0 0 IN IP4 67.150.160.10 s=sip call c=IN IP4 67.150.160.10 t=0 0 m=audio 17308 RTP/AVP 0 <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<< IN <<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< SIP/2.0 200 OK Via: SIP/2.0/UDP X.X.X.90;branch=z9hG4bK2aa.c3961912.0 Via: SIP/2.0/UDP X.X.X.238:5060;received=X.X.X.238;branch=z9hG4bK10281a99;rport=5060 Record-Route: <sip:X.X.X.90;lr;ftag=as4c62e0b9> From: ""2026097878"" <sip:2026097878@mydomain.com<sip%3A2026097878(a)mydomain.com> >;tag=as4c62e0b9 To: <sip:+16782247008@X.X.X.90:5060>;tag=H3jDp3KZmHNja Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com CSeq: 102 INVITE Contact: <sip:16782247008@67.150.160.10 <sip%3A16782247008(a)67.150.160.10>> Content-Type: application/sdp Content-Length: 109 v=0 o=ATSI-B1 0 0 IN IP4 67.150.160.10 s=sip call c=IN IP4 67.150.160.10 t=0 0 m=audio 17308 RTP/AVP 0 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>> OUT >>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> ACK sip:16782247008@67.150.160.10 <sip%3A16782247008(a)67.150.160.10> SIP/2.0 Via: SIP/2.0/UDP X.X.X.238:5060;branch=z9hG4bK60b74898;rport Route: <sip:X.X.X.90;lr;ftag=as4c62e0b9> From: ""2026097878"" <sip:2026097878@mydomain.com<sip%3A2026097878(a)mydomain.com> >;tag=as4c62e0b9 To: <sip:+16782247008@X.X.X.90:5060>;tag=H3jDp3KZmHNja Contact: <sip:2026097878@X.X.X.238> Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com CSeq: 102 ACK User-Agent: G-Tel v1.0 Max-Forwards: 70 Content-Length: 0 <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<< IN <<<<<<<<<<<<<<<<<<<<<<<< <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< BYE sip:X.X.X.90;lr;ftag=as4c62e0b9 SIP/2.0 Via: SIP/2.0/UDP 67.150.160.10;branch=z9hG4bKvNBHaK6ZByQmg Max-Forwards: 70 From: <sip:+16782247008@X.X.X.90:5060>;tag=H3jDp3KZmHNja To: ""2026097878"" <sip:2026097878@mydomain.com<sip%3A2026097878(a)mydomain.com> >;tag=as4c62e0b9 Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com CSeq: 102 BYE Content-Length: 0 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>> OUT >>>>>>>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> SIP/2.0 404 Not here Via: SIP/2.0/UDP 67.150.160.10;branch=z9hG4bKvNBHaK6ZByQmg From: <sip:+16782247008@X.X.X.90:5060>;tag=H3jDp3KZmHNja To: ""2026097878"" <sip:2026097878@mydomain.com<sip%3A2026097878(a)mydomain.com> >;tag=as4c62e0b9 Call-ID: 4199b75502736b01448ce4bf04876d7e(a)mydomain.com CSeq: 102 BYE Server: Kamailio Content-Length: 0 [View Less]

2 1

failure_route does not fire up after
by Mino Haluz 12 Oct '10

12 Oct '10

Hi, I've been struggling with this problem very long time and I am starting to be quite desperate. I'm trying to implement voicemail redirection after 15 secs of ringing. So I set fr_inv_timer, set the event handler with t_on_failure. But after 15secs of ringing the failture_route block does not execute at all, the transaction is lost and after these 15 secs it has strange behaviour (the phone is still ringing, even after the other side has cancelled the call). I'm attaching the … [View More]configuration without sensitive data. Please tell me whether problem is in the configuration or somewhere outside of kamailio. Thanks for any suggestion. #!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 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 dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_call ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif # ----------------------------------------------------------------------------------------- #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_NAT # ----------------------------------------------------------------------------------------- ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL7 fork=yes #fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* 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:1.1.1.1:5060 # ----------------------------------------------------------------------------------------- alias=test # ----------------------------------------------------------------------------------------- port=5060 ####### 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: $sel(cfg_get.group.id) or @cfg_get.group.id # ####### Modules Section ######## #set module path #mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" mpath="/lib/kamailio/modules_k/:/lib/kamailio/modules/" /* uncomment next line for MySQL DB support */ #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" # Addition modules # ----------------------------------------------------------------------------------------- loadmodule "avpops.so" loadmodule "regex.so" loadmodule "sqlops.so" loadmodule "dialog.so" loadmodule "call_control.so" loadmodule "mediaproxy.so" loadmodule "siptrace.so" loadmodule "mi_datagram.so" loadmodule "group.so" loadmodule "permissions.so" # ----------------------------------------------------------------------------------------- modparam("mi_datagram", "socket_name", "/var/run/kamailio/kamailio.sock") # The name of a UNIX SOCKET used by Callcontrol modparam("dialog", "dlg_flag", 4) #modparam("call_control", "disable", 0) #modparam("call_control", "signaling_ip_avp", "$avp(s:signaling_ip)") # ----------------------------------------------------------------------------------------- modparam("sqlops","sqlcon","ca=>mysql:/") # The definition of a DB connection modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy/dispatcher.sock") # Path to the filesystem socket where the mediaproxy dispatcher listens for commands # ----------------------------------------------------------------------------------------- modparam("tm", "ruri_matching", 1) modparam("tm", "restart_fr_on_each_reply", 0) # Final response timer will be re-triggered for each received provisional reply #modparam("tm", "fr_timer", 15000) # Timer which hits if no final reply for an INVITE arrives (15 sec) modparam("tm", "fr_inv_timer", 15000) # Timer which hits if no final reply for an INVITE arrives (15 sec) # ----------------------------------------------------------------------------------------- modparam("siptrace", "db_url", "") # Database URL modparam("siptrace", "traced_user_avp", "$avp(s:traced_user)") modparam("siptrace", "trace_on", 1) modparam("siptrace", "trace_flag", 22) # Flag is used to mark messages to trace # ----------------------------------------------------------------------------------------- # Radius settings modparam("acc", "failed_transaction_flag", 1) modparam("acc", "report_cancels", 0) modparam("acc", "report_ack", 0) modparam("acc", "early_media", 0) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 1) modparam("acc", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") # This is the location of the configuration file of radius client modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 1) modparam("acc", "radius_extra", "User-Name=$Au; \ Calling-Station-Id=$from; \ Called-Station-Id=$to; \ Sip-Translated-Request-URI=$avp(s:translated_uri); \ Sip-RPid=$avp(s:rpid); \ Source-IP=$si; \ Source-Port=$sp; \ Canonical-URI=$avp(s:can_uri); \ Billing-Party=$avp(s:billing_party); \ Divert-Reason=$avp(s:divert_reason); \ X-RTP-Stat=$hdr(X-RTP-Stat); \ Contact=$hdr(contact); \ Event=$hdr(event); \ SIP-Proxy-IP=$avp(s:sip_proxy_ip); \ ENUM-TLD=$avp(s:enum_tld)") # ----------------------------------------------------------------------------------------- #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!endif /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ loadmodule "domain.so" #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" #!endif loadmodule "uac.so" # ----------------- setting module-specific parameters --------------- # --- enable CLI with append_rpid_hf --- modparam("auth", "rpid_prefix", "<sip:") modparam("auth", "rpid_suffix", "@1.1.1.1;user=phone>;party=calling; screen=yes; privacy=off") #modparam("auth", "rpid_avp", "$avp(s:rpid)") # ----- mi_fifo params ----- # ----------------------------------------------------------------------------------------- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----------------------------------------------------------------------------------------- # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 1) # ----- rr params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "mysql://") modparam("domain", "db_url", "mysql://") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "db_url", "mysql://") modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://") #!endif # ----- auth_db params ----- /* enable the DB based authentication */ #!ifdef WITH_AUTH modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://") modparam("auth_db", "load_credentials", "rpid") #!endif # -- group params -- /* for grp mysql table usage - groups - siptrace, blocked*/ modparam("group", "db_url", "mysql://") modparam("group", "table", "grp") modparam("group", "user_column", "username") modparam("group", "domain_column", "domain") modparam("group", "group_column", "grp") # -- permissions params -- /* for trusted table usage and other functions - other modparam are default */ modparam("permissions", "db_url", "mysql://") modparam("permissions", "db_mode", 1) # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ #modparam("domain", "db_url", # "mysql://") #modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* enable presence server support */ #!ifdef WITH_PRESENCE modparam("presence|presence_xml", "db_url", "mysql://") modparam("presence_xml", "force_active", 1) modparam("presence", "server_address", "sip:10.0.0.10:5060") #!endif # ----- nathelper ----- #!ifdef WITH_NAT #modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") #modparam("nathelper", "natping_interval", 30) #modparam("nathelper", "ping_nated_only", 1) #modparam("nathelper", "sipping_bflag", 7) #modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org<sip%3Apinger(a)kamailio.org> ") modparam("registrar|nathelper", "received_avp", "$avp(i:80)") modparam("usrloc", "nat_bflag", 6) #!endif ####### Routing Logic ######## # main request routing logic route{ #------------SANITY CHECK---------------------------------------------------------------- xlog("L_INFO","XLOG ****************START*($rm)*S_IP($si)*CallID($ci)*********************"); if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); xlog("L_INFO","XLOG: Too many hops man"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } if(!sanity_check("1511", "7")) { xlog("L_INFO","XLOG: Malformed SIP message"); xlog("Malformed SIP message from $si:$sp\n"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } # NAT detection xlog("L_INFO", "XLOG: Performing NAT detection"); route(NAT); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { xlog("L_INFO", "XLOG: loose_route()"); if (is_method("BYE")) { xlog('L_INFO', 'XLOG: Method is BYE'); setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails # end_media_session(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { xlog('L_INFO', 'XLOG: Method is SUBSCRIBE and uri == myself'); # in-dialog subscribe requests route(PRESENCE); #!ifndef WITH_PRESENCE #added by yenkee sl_send_reply("403", "Not Allowed"); #!endif xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } if ( is_method("ACK") ) { xlog('L_INFO', 'XLOG: Method is ACK'); if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server xlog('L_INFO', 'XLOG: t_check_trans()'); t_relay(); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } else { xlog('L_INFO', 'XLOG: ACK without matching transaction ... ignore and discard.'); #added by yenkee sl_send_reply("403", "Not Allowed"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } } sl_send_reply("404","Not here"); xlog("L_INFO", "XLOG: 404 Not Here ----------------------------------"); } xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { xlog('L_INFO', 'XLOG: Method is CANCEL'); if (t_check_trans()) t_relay(); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } t_check_trans(); # authentication route(AUTH); setflag(1); # 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")) { xlog("L_INFO","XLOG: Method is INVITE"); setflag(1); # do accounting # ----------------------------------------------------------------------------------------- $avp(s:can_uri) = $ru; # Canonical-URI will be used for rating the session xlog("L_INFO", "XLOG: Engaging Media Proxy"); engage_media_proxy(); # Automatically call use_media_proxy / end_media_proxy on every request and reply that belongs to the dialog # ----------------------------------------------------------------------------------------- } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { xlog("L_INFO","XLOG: Appeding Hf P-hing: outbound"); append_hf("P-hint: outbound\r\n"); route(RELAY); } # requests for my domain if( is_method("PUBLISH|SUBSCRIBE")) route(PRESENCE); if (is_method("REGISTER")) { if(isflagset(5)) { setbflag("6"); # uncomment next line to do SIP NAT pinging ## setbflag("7"); } if (!save("location")) sl_reply_error(); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } # ---------ENABLING SIP TRACE-------------------------------------------------------------- # Enable sip tracing for users in grp siptrace in mysql db ( nema vplyv na zapisovanie acc informacii ) if (is_user_in("From", "siptrace") ) { xlog("L_INFO","XLOG: Starting sip_trace for From-URI:($fu)"); setflag(22); # sip_trace(); # mne sip_trace nefungoval iba setflag , ak sa nastavi flag na hodnotu v modparam bude robit cdr tool sip tracy } # route(PSTN); # ----------------------------------------------------------------------------------------- # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if (is_method("INVITE")) { #------------BLOCK NUMBER----------------------------------------------------------------- # Block username if ( is_user_in("From", "blocked")) { xlog("L_INFO", "XLOG: [main route] Account ($fu) to ($ru) is blocked"); sl_send_reply("403", " Account blocked "); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } # ----------------------------------------------------------------------------------------- # Call control allows one to limit the duration of calls and automatically end them when they exceed the imposed limit xlog("L_INFO","XLOG: Before call_control() var rm = ($rm)"); call_control(); switch ($retcode) { case 2: # Call with no limit xlog("L_INFO", "XLOG: [call_control] Call with no limit"); break; case 1: # Call with a limit under callcontrol management (either prepaid or postpaid) xlog("L_INFO", "XLOG: [call_control] Call with a limit under callcontrol management"); break; case -1: # Not enough credit (prepaid call) xlog("L_INFO", "XLOG: Call control: not enough credit for prepaid call"); acc_rad_request("402"); sl_send_reply("402", "Not enough credit"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; break; case -2: # Locked by call in progress (prepaid call) xlog("L_INFO", "XLOG: Call control: prepaid call locked by another call in progress"); acc_rad_request("403"); sl_send_reply("403", "Call locked by another call in progress"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; break; default: # Internal error (message parsing, communication, ...) xlog("L_INFO", "XLOG: Call control: internal server error"); acc_rad_request("500"); sl_send_reply("500", "Internal server error"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } # ----------------------------------------------------------------------------------------- # ----------------------------------------------------------------------------------------- # Return error message when you are calling on your own number # fU - From username # tU - To username if ($fU==$tU) { sl_send_reply("403","Forbidden calling own number"); xlog("L_INFO","XLOG: Call on your own number: Request-URI:($ru) From-URI:($fu). Sending 403 Forbidden"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } # ----------------------------------------------------------------------------------------- } # ----------------------------------------------------------------------------------------- # ----------------------------------------------------------------------------------------- # ----------------------------------------------------------------------------------------- # ----------------------------------------------------------------------------------------- xlog("L_INFO","XLOG: Performing lookup( location) "); if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "[local failed] Not Found"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; break; # commented - not used with PSTN case -2: sl_send_reply("405", "Method Not Allowed"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } # ----------------------------------------------------------------------------------------- # ----------------------------------------------------------------------------------------- # ----------------------------------------------------------------------------------------- # Replacing plus sign with 00 if ($ru=~".*sip:\+.*@.*") { # Check if URI contains plus sign $avp(s:to)=$ru; avp_subst("$avp(s:to)", "/(sip:)\+/sip:00/"); # Replace plus sign with 00 avp_pushto("$ru", "$avp(s:to)"); # Push the value of AVP with name into the SIP message as Request-URI xlog("L_INFO","XLOG: Replacing plus sign in Request-URI ($ru)"); } # ----------------------------------------------------------------------------------------- # ----------------------------------------------------------------------------------------- } # when routing via usrloc, log the missed calls also setflag(2); route(RELAY); } route[RELAY] { xlog("L_INFO", "XLOG: [RELAY] Jumped to route(RELAY)"); #!ifdef WITH_NAT if (check_route_param("nat=yes")) { xlog("L_INFO","XLOG: [RELAY] check_route_param('nat=yes') == 1"); setbflag("6"); } if (isflagset(5) || isbflagset("6")) { xlog("L_INFO","XLOG: [RELAY] isflagset(5) && isbflagset(6)"); route(RTPPROXY); } #!endif # ----------------------------------------------------------------------------------------- /* example how to enable some additional event routes */ if (is_method("INVITE")) { # $avp(s:callee_fr_inv_timer) = 10; xlog("L_INFO", "XLOG: [RELAY] Routing of failure and reply messages"); # t_on_branch("BRANCH_ONE"); t_set_fr(10000); t_on_reply("REPLY_ONE"); t_on_failure("1"); # Registering failure route } if (!t_relay()) { sl_reply_error(); } xlog("L_INFO","XLOG: exiting FROM RELAY ------------------------------------"); exit; } # Presence server route route[PRESENCE] { #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } return; } # Authentication route route[AUTH] { xlog("L_INFO", "XLOG: Jumped to route[AUTH]"); #!ifdef WITH_AUTH if (is_method("REGISTER")) { xlog("L_INFO","XLOG: [AUTH] Register pred ifom From-URI:($fu)"); # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("", "subscriber")) { xlog("L_INFO", "XLOG: [AUTH] www_challenge()"); www_challenge("", "0"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } if ($au!=$tU) { xlog("L_INFO", "XLOG: [AUTH] Forbidden auth ID From-URI:($fu)"); sl_send_reply("403","Forbidden auth ID"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } } else { # authenticate if from local subscriber (uncomment to enable auth) if (from_uri==myself) { xlog("L_INFO","XLOG: [AUTH] Message is not REGISTER, and from_uri == myself"); if (!proxy_authorize("", "subscriber")) { xlog("L_INFO", "XLOG: [AUTH] proxy_challenge()"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); proxy_challenge("", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } } xlog("L_INFO","XLOG: [AUTH] consumer_credentials()"); consume_credentials(); # caller authenticated } } #!endif return; } # Caller NAT detection route route[NAT]{ #!ifdef WITH_NAT force_rport(); xlog("L_INFO", "XLOG: Checking nat_uac_test('19')"); if (nat_uac_test("19")) { if (method=="REGISTER") { xlog("L_INFO", "XLOG: fix_nated_register()"); fix_nated_register(); } else { xlog("L_INFO", "XLOG: fix_nated_contact()"); fix_nated_contact(); } setflag(5); } #!endif return; } # RTPProxy control route[RTPPROXY] { xlog("L_INFO","XLOG: Jumped to route[RTPPROXY]"); #!ifdef WITH_NAT # if (is_method("BYE")) { # unforce_rtp_proxy(); # } else if (is_method("INVITE")){ # force_rtp_proxy(); # } if (!has_totag()) { add_rr_param(";nat=yes"); xlog("L_INFO","XLOG: Adding RR param(';nat=yes')"); } #!endif return; } # 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"); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; #!endif return; } # Sample branch router branch_route[BRANCH_ONE] { xlog("L_INFO","XLOG: BRANCH ROUTE "); exit; } # Sample onreply route onreply_route[REPLY_ONE] { #xdbg("incoming reply\n"); xlog("L_INFO","XLOG: REPLY ROUTE "); #!ifdef WITH_NAT # if ((isflagset(5) || isbflagset("6")) && status=~"(183)|(2[0-9][0-9])") { # force_rtp_proxy(); # } if (isbflagset("6")) { fix_nated_contact(); } #!endif } # Sample failure route failure_route[1] { #!ifdef WITH_NAT # if (is_method("INVITE") # && (isbflagset("6") || isflagset(5))) { # unforce_rtp_proxy(); # } #!endif xlog("L_INFO","XLOG ****************FAILROUTE*($rm)*S_IP($si)**CallID($ci)********************"); if (t_is_canceled()) { xlog("L_INFO","XLOG: exiting ----------------------------------------"); exit; } # ----------------------------------------------------------------------------------------- # Forward call on Voicemail when spefified SIP Status Code is returned. Failure route registered using t_on_failure. # 486 - Busy Here # 408 - Request Timeout # 480 Temporarily Unavailable # if (t_check_status("486|408|480")) { if (t_check_status("486|408")) { xlog("L_INFO","XLOG: [failure route] Forwarding on Voicemail"); } # ----------------------------------------------------------------------------------------- } [View Less]

2 1

Caller ID issue
by Lucas Alvarez 12 Oct '10

12 Oct '10

Hi, I'm having a problem with the caller ID, I have implemented an integration between asterisk and kamailio following this tutorial: http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb and the problem is that when I call from extension, let's say 1000, to another extension, let's say 2000, the callerid number is always the number I'm calling, in this case 2000. Using xlog and printing $fu, $fU variables I realize that when the call came from asterisk to the destination … [View More]

2 3

Stateful vs. stateless replies from script
by Alex Balashov 12 Oct '10

12 Oct '10

Now that K v3.1.0 provides send_reply(), what is the preferred ideology about whether to send stateful or stateless negative error replies? I mean in general, not in specific cases like digest authentication in the new 'auth' module, where, according to the docs, a transaction and stateful replies are required in order for enhanced nonce replay protection. The traditional wisdom in past bodies of scripts and knowledge in the public realm has been, in my interpretation, that stateless … [View More]

3 6

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sr-users October 2010