sr-users August 2007

sr-users@lists.kamailio.org

169 participants
286 discussions

by Daniel-Constantin Mierla

Hello, courtesy of Glenn Marmon, Connecticut College Class of 2007, who submitted the patch, OpenSER includes now support for SCTP (Stream Control Transmission Protocol). It completes the list of the possible IP transport layers that could be used with SIP in OpenSER. You can find details about SCTP at: http://www.sctp.org/ Basically, it is a reliable transport protocol, like TCP, but provides a number of functions that are critical for telephony signaling transport (read more about the benefits: http://www.ietf.org/rfc/rfc3286.txt). If you have the chance to own a SCTP-aware SIP phone and can test, please provide us feedback on the mailing lists -- also, let us know the name and type of any SIP phone with SCTP support that you know. Hope SCTP will gain more and more space, as it seems to be the right protocol for VoIP/IM&P signaling, where UDP cannot cope very well with huge messages and TCP adds too much overhead. Cheers, Daniel

17 years, 3 months

[OpenSER-Users] Logs to /var/log/openser

by Iñaki Baz Castillo

Hi, I've configured: log_facility=LOG_LOCAL0 and added in "syslog.conf": # don't log messages with LOG_LOCAL0 in /var/log/syslog anymore *.*;auth,authpriv.none,local0.none -/var/log/syslog # # log messages with LOG_LOCAL0 in /var/log/ser.log local0.* -/var/log/ser.log as it appears in http://docs.huihoo.com/openser/tutorials/ser-syslog/ar01s06.html OpenSer then logs in /var/log/openser, true, but it still logs in /var/log/syslog. ¿?¿ Obviously I don't want OpenSer logs appearing in /var/log/syslog. I use Debian Etch, any idea of what can be wrong? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

17 years, 3 months

[OpenSER-Users] avp_subst and regex

by Zahid Mehmood

Hi, I am trying to rewrite a 5 digit number to a 11 digit number by adding a prefix "121285". To achieve that I use the following: avp_subst("$avp(s:fromuri)","/^(sip:)([134][0-9]{4}@.*)/\1+121285\2/"); in openser.log new fromuri = sip:+12128512345@host.columbia.edu It works fine because of the "+" after \1 This avp_subst does not work properly if the "+" is taken out. How can I use "\1" followed by another digit? avp_subst("$avp(s:fromuri)","/^(sip:)([134][0-9]{4}@.*)/\1121285\2/"); in openser.log: new fromuri = J128512345(a)host.columbia.edu What am i doing wrong? Thanks in advance for your help. -- Zahid

17 years, 3 months

[OpenSER-Users] OpenSER looping routes back to itself.

by Taylor Carpenter

Running OpenSER v1.1.1 with internal ip for sip client connections and ext ip to connect to sip providers. I am getting loops in my calls sometimes... They stop after reaching the max hops trying to send the BYE message to it self. I am wondering if it is the record- route that I see that includes the internal IP. The 483 error is U 2007/07/30 00:49:56.133157 192.168.18.22:5060 -> 192.168.18.22:5060 SIP/2.0 483 Too Many Hops. From: <sip:15126461516@192.168.18.1:5060>;tag=nu-481d-6c62. To: <sip:18065436468@192.168.18.22:5060>;tag=6738. Call-ID: 421ac0c6-1dd2-11b2-8093-df998124da20(a)192.168.18.91. CSeq: 11269143 BYE. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.7fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.6fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.5fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.4fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.3fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.2fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.1fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.0fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.fee832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.eee832a5.0. Via: SIP/2.0/UDP 192.168.18.91:5060;rport=5060. Server: OpenSer (1.1.1-notls (i386/linux)). Content-Length: 0. Any ideas?

17 years, 3 months

[OpenSER-Users] benchmark module

by Daniel-Constantin Mierla

Hello, starting from the original patch submitted by Bastian Friedrich and incorporating the code used in the core for past benchmarking, the devel version has a new module "benchmark". It can be used to log the duration of a configuration part execution, allowing in this way to detect slow elements of routing script. I am sure is a important tool, at least in preparing production systems, so development will continue to make more flexible the printed statistics. The readme file will give you more details: http://www.openser.org/docs/modules/1.3.x/benchmark.html The module exports an internal API, so developers can do benchmarking inside the C code. Feedback is appreciated. Cheers, Daniel

17 years, 3 months

Re: [OpenSER-Users] Replacing "." of a domain with "_" and store the result in a variable

by Iñaki Baz Castillo

El Thursday 02 August 2007 17:31:23 escribió: > Hi, > > $avp(s:dom) = $rd; > avp_subst("$avp(s:dom)", "/\./_/g"); Great, didn't realize of that :) Thanks a lot. > > Andreas > > Iñaki Baz Castillo wrote: > > Hi, I just need to take the requested domain $rd and store in a variable > > that domain but with . replaced by _: > > > > sip.domain.org -> sip_domain_org > > > > This is because I want tables in MySQL containing the domain name but > > MySQL don't allow . in table name. > > > > So now, what should I do for that? I really thought "textops" could help > > me but I don't see a function to replace text and store it in a variable. > > > > I see "replace()" and "subst()" but they seem to exist for replacing text > > in the SIP message "on the fly", that is not that I'm looking for. > > > > Any suggestion? Thanks a lot. -- ilimit... *Iñaki Baz Castillo* ibc(a)in.ilimit.es ÀREA SISTEMES 0034 937 333 375 VOLTA 1, PIS 5 08224 TERRASSA.BCN Aquest enviament és confidencial i està destinat únicament a la persona a qui s'ha enviat. Pot contenir informació privada sotmesa al secret professional, la distribució de la qual està prohibida per la legislació vigent.

17 years, 3 months

[OpenSER-Users] Replacing "." of a domain with "_" and store the result in a variable

by Iñaki Baz Castillo

Hi, I just need to take the requested domain $rd and store in a variable that domain but with . replaced by _: sip.domain.org -> sip_domain_org This is because I want tables in MySQL containing the domain name but MySQL don't allow . in table name. So now, what should I do for that? I really thought "textops" could help me but I don't see a function to replace text and store it in a variable. I see "replace()" and "subst()" but they seem to exist for replacing text in the SIP message "on the fly", that is not that I'm looking for. Any suggestion? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

17 years, 3 months

[OpenSER-Users] OPENSERAMIN:SocketError in loginController#login

by zze-WU Yin RD-MAPS-LAN

Hello everybody, I haved installed a openseradmin. And I can have a page for login in the http://localhost:3000. But when i loggined with: login: openser password:openserrw I have got a page of error : SocketError in loginController # login Do anybody know why? Thank you very much! Yin

17 years, 3 months

[OpenSER-Users] NAT + STUN

by Marc LEURENT

Hello evrybody, I need a piece of advice if you have time.... Thanks I have some trouble with my configuration file, I'm trying to set a SIP network working through NAT using STUN. REGISTRATION is working, INVITE are working, I can set a communication with 2 working phones but sip pings an BYE messages are not forwared... Any idea?? Thanks # U 88.191.45.91:5060 -> 82.127.0.79:1312 OPTIONS sip:82.127.0.79:1312 SIP/2.0. Via: SIP/2.0/UDP 88.191.45.91:5060;branch=0. From: sip:pinger@sip.leurent.eu;tag=804466c1. To: sip:82.127.0.79:1312. Call-ID: 23ce8297-c09c8eb7-a(a)88.191.45.91. CSeq: 1 OPTIONS. Content-Length: 0. . # U 82.127.0.79:1312 -> 88.191.45.91:5060 SIP/2.0 481 CallLeg/Transaction Does Not Exist. Via: SIP/2.0/UDP 88.191.45.91:5060;branch=0. From: <sip:pinger@sip.leurent.eu>;tag=804466c1. To: <sip:82.127.0.79:1312>;tag=c0a80101-4bf1d8. Call-ID: 23ce8297-c09c8eb7-a(a)88.191.45.91. CSeq: 1 OPTIONS. Content-Length: 0. . # U 82.127.0.79:1312 -> 88.191.45.91:5060 BYE sip:103@82.127.0.79:1027 SIP/2.0. Via: SIP/2.0/UDP 82.127.0.79:1313;branch=z9hG4bK8030359792092547043. From: "101"<sip:101@sip.leurent.eu:5060;user=phone>;tag=c0a80101-4c5eed. To: <sip:103@sip.leurent.eu:5060;user=phone>;tag=c0a80101-1d0bb0d. Call-ID: 66464a0-c0a80101-0-1f(a)192.168.95.4. CSeq: 2 BYE. Max-Forwards: 70. Route: <sip:88.191.45.91:5060;lr=on;ftag=c0a80101-4c5eed>. User-Agent: THOMSON ST2030 hw0 fw1.50 00-0E-50-4E-AF-AE. Content-Length: 0. . # # $Id: openser.cfg 1827 2007-03-12 15:22:53Z bogdan_iancu $ # # simple quick-start config script # Please refer to the Core CookBook at http://www.openser.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # # ----------- global configuration parameters ------------------------ debug=7 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) children=4 #alias=sip.leurent.eu port=5060 #server_signature=yes #tos=IPTOS_LOWDELAY avp_aliases="day=i:101;time=i:102;can_uri=i:800;s_ip=i:801;billing_party=i:802;from_header=i:803;sip_proxy_ip=i:804" #;pstnuser=i:805;pstnpassword=i:806:pstnrealm=i:807" # ------------------ module loading ---------------------------------- #set module path mpath="/usr/lib/openser/modules/" # Uncomment this if you want to use SQL database loadmodule "mysql.so" loadmodule "sl.so" # Stateless Module loadmodule "tm.so" # Transaction Module loadmodule "rr.so" # Record-Route and Route Module loadmodule "maxfwd.so" # Max-Forward processor Module loadmodule "usrloc.so" # User Location Implementation Module loadmodule "registrar.so" # SIP Registrat Implementation Module (need usrloc) loadmodule "textops.so" # Text Operation Module loadmodule "mi_fifo.so" # FIFO transport layer implementation for Management Interface loadmodule "acc.so" # Accounting Module loadmodule "avpops.so" # AVP Operation Module (user preference) loadmodule "uri.so" # Generic URI operation Module loadmodule "auth.so" # Authentification Module #loadmodule "auth_db.so" # Database-backend Authentication mMdule loadmodule "auth_radius.so" # RADIUS-backend Authentication Module loadmodule "group_radius.so" # User-groups Module with RADIUS-backend #loadmodule "avp_radius.so" # RADIUS-backend for AVP loading Module #loadmodule "presence.so" # Presence server Module #loadmodule "pua.so" # Common API for presence user agent client loadmodule "options.so" # OPTIONS server replier Module loadmodule "xlog.so" # Advanced Logger Module loadmodule "nathelper.so" # NAT Traversal Helper Module #loadmodule "dispatcher.so" # Dispatcher (load-balancer) Module loadmodule "uac.so" # User Agent Client loadmodule "siptrace.so" # SipTrace module (storage of SIP requests) #loadmodule "exec.so" # Allows to start an external command from a OpenSER script # ----------------- setting module-specific parameters --------------- # -- maxfwd params -- modparam("maxfwd", "max_limit", 10) # Default is 256 | 10 in the functions # -- sl params -- #modparam("sl", "enable_stats", 1) # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- # Uncomment this if you want to use SQL database modparam("usrloc", "db_mode", 1) # Write instantaneously in the DB modparam("usrloc", "db_url", "mysql://openser:test@127.0.0.1/openser") modparam("usrloc", "timer_interval", 10) modparam("usrloc", "nat_bflag" , 3) # -- rr params -- modparam("rr", "enable_full_lr", 1) # add value to ;lr param to make some broken UAs happy # -- siptrace params -- modparam("siptrace", "db_url", "mysql://openser:test@127.0.0.1/openser") modparam("siptrace", "table", "sip_trace") # Default value "sip_trace" modparam("siptrace", "trace_on", 1) # -- registrar params -- modparam("registrar", "default_expires", 1800) modparam("registrar", "received_avp", "$avp(i:42)") # -- nathelper params -- modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "sipping_bflag", 5) modparam("nathelper", "natping_interval", 10) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "received_avp", "$avp(i:42)") # Same Value as the registrar module modparam("nathelper", "sipping_from", "sip:pinger@sip.leurent.eu") # -- auth params -- #modparam("auth", "secret", "johndoessecretphrase") # Default is random => don't set it #modparam("auth", "nonce_expire", 300) # Time before nounce expiration modparam("auth_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") # -- group_radius params -- modparam("group_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("group_radius", "use_domain", 1) # username@domain will be used for lookup # -- avp_radius parameter -- #modparam("avp_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") # -- acc params (with radius )-- modparam("acc", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) modparam("acc", "early_media", 1) modparam("acc", "report_cancels", 1) #modparam("acc", "report_ack", 0) modparam("acc", "detect_direction", 1) #modparam("acc", "log_flag", 1) # number of the flag which will be used to mark messages for accounting #modparam("acc", "log_level", 1) # Set the reporting log level #modparam("acc", "log_missed_flag", 2) # #modparam("acc", "failed_transaction_flag", 2) modparam("acc", "service_type", 15) # Radius service type used for accounting : 15 = (SIP) #modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp") # ATTENTION: DO NOT PUT ; at the end of the radius_extra attribute modparam("acc", "radius_extra", "Sip-Src-IP=$si; Sip-Src-Port=$sp; Canonical-URI=$avp(can_uri); Billing-Party=$avp(billing_party); SIP-Proxy-IP=$avp(sip_proxy_ip); User-Agent=$ua ") #Billing-Party=$avp(billing_party) #From-Header=$hdr(from); #User-Name=$fU; #From-Header=$avp(from_header); #Digest-Realm=$fd #Sip-From-Tag=$avp(from_header); #SIP-Method=$rm; # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # NAT detection route(2); # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") { record_route(); }; # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); if(is_method("BYE")) { # log it all the time acc_rad_request("200 ok"); acc_log_request("200 ok"); } route(1); }; # Set the acc flags if(is_method("INVITE") && !has_totag()) { xlog("L_INFO", "I AM SETTING THE FLAGS FOR RADIUS \r\n"); $avp(can_uri) = $ru; # SIP Request's URI $avp(billing_party) = $fu; # From URI $avp(from_header) = $fU; # From URI username $avp(sip_proxy_ip) = $Ri; # Received IP address setflag(1); # radius_flag setflag(2); # radius_missed_flag }; # Functions when calling other domains if (!uri==myself) { # check if user is allowed to do voip calls to other domains # if(is_method("INVITE|MESSAGE")) { # if (radius_is_user_in("From", "voip")) { # sl_send_reply("403", "Forbidden VoIP"); # exit; # }; # }; # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { sip_trace(); xlog("L_INFO", "$fU IS TRYING TO REGISTER \r\n"); if (!radius_www_authorize("sip.leurent.eu")) { www_challenge("sip.leurent.eu", "0"); # qop set to 1 xlog("L_INFO", "WWW_CHALLENGE of $si FAILED \r\n"); exit; }; #if (isflagset(5)) { if (isbflagset(3)) { #setflag(6); # if you want OPTIONS natpings uncomment next # setflag(7); # Deprecated setbflag(5); # Set Flag for SIP PINGING }; save("location"); xlog("L_INFO", "SAVE LOCATION OF $si \r\n"); exit; }; if (!lookup("location")) { # log to acc as missed call acc_rad_request("404 Not Found"); acc_log_request("404 Not Found"); xlog("L_DBG", "ACC RADIUS: 404 NOT FOUND FOR $si \r\n"); sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } ## Generic Forward route[1] { if (subst_uri('/(sip:.*);nat=yes/\1/')){ #setflag(6); # Deprecated, for version 1.1 setbflag(3); }; #if (isflagset(5)||isflagset(6)) { if (isbflagset(3)) { route(3); } if (!t_relay()) { sl_reply_error(); }; exit; } # NAT Detection route[2]{ force_rport(); if (nat_uac_test("19")) { xlog("!!!!!!!!! NAT UAC TEST 19 SUCEDEED \r\n"); if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; #setflag(5); Deprecated setbflag(3); }; } ## Route for natted contact route[3] { if (is_method("BYE|CANCEL")) { # Ajout Maison acc_rad_request("200 ok"); acc_log_request("200 ok"); #unforce_rtp_proxy(); } else if (is_method("INVITE")){ #force_rtp_proxy(); t_on_failure("1"); }; #if (isflagset(5)) if (isbflagset(3)){ search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); } t_on_reply("1"); } ## Failure Route 1 failure_route[1] { xlog("!!!!!!!!! ON FAILURE ROUTE \r\n"); #if (isflagset(6) || isflagset(5)) { if (isbflagset(3)) { #unforce_rtp_proxy(); } } ## Reply route onreply_route[1] { xlog("!!!!!!!!! ON REPLY ROUTE \r\n"); #if ((isflagset(5) || isflagset(6)) && status=~"(183)|(2[0-9][0-9])") { if (isbflagset(3) && status=~"(183)|(2[0-9][0-9])") { #force_rtp_proxy(); } search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); #if (isflagset(6)) { if (isbflagset(3)) { xlog("!!!!!!!!! ON REPLY ROUTE / FIX NATED CONTACT \r\n"); fix_nated_contact(); } exit; }

17 years, 3 months

[OpenSER-Users] openser to solve SIP+RTP+NAT problems in typical SOHO setup

by Jens Thiele

Hi, I am evaluating software to solve SIP+RTP+NAT problems in typical SOHO setups: Scenarios: Typical scenarios are (A)DSL "dial-up / dynamic IP" connections with one public IP and a router doing NAT for some LANs. 1) WAN multiple LANs internet <-> NAT router <-> SIP phones sip registrar SIP proxy <-> SIP phones with NAT logic : 2) WAN multiple LANs internet <-> NAT router <-> SIP proxy with NAT logic, SIP phones sip registrar <-> SIP phones : Notes: - the SIP phones use the SIP proxy with NAT logic as outbound proxy - the proxy is notified if the public IP changes The closest match so far has been siproxd [1]. But I would prefer to use openser+nathelper+rtpproxy. Now it seems nathelper is designed for a different scenario (SIP provider): WAN/internet sip registrar <-> ser+nathelper <-> NAT router <-> multiple LANs with SIP phones and uses the packet source address in fix_nated_contact. Now I stumbled across milkfish [2] and its patches [3] to openser (nathelper). It seems the patch allows to explicitly set the address used by fix_nated_contact to rewrite the contact header (instead of using the packet source address). Searching the mailing list archives I did not find a better solution. Closest match I found was the thread: "[Devel] Extended nathelper" started by Walter Schober http://www.mail-archive.com/devel@openser.org/msg01930.html Ending in the suggestion to use add_rvc_param(). But add_rcv_param seems to use the packet source address, too. But also from this thread it seems one might use the mangle module to rewrite the contact header field, but then I am not sure about the state of this module: "This is a module to help with SDP mangling. Still in testing." I am a bit surprised that there is no straight forward solution to such a common setup. Am I missing something really obvious? Thanks Greetings Jens Thiele [1] siproxd <http://siproxd.sourceforge.net/> [2] milkfish <http://wiki.milkfish.org> [3] milkfish openser patches <http://packages.milkfish.org/boozy/Milkfish_Sources_for_OpenWrt-SDK/OpenWrt…>

17 years, 3 months

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sr-users August 2007