Iñaki, Thank you for your response. I have enabled the siptrace module in openser. The data in the mysql table only shows the trace between the carrier and openser. Can I submit a pcap file that shows all of the SIP communication that occured during the call. Thank You Stagg Shelton

Thanks again Iñaki. I am attaching siptrace.txt file. I can see that there appears to be something odd with the ACKs in that they appear to be sent from my openser back to my openser in a loop until the max forwards is reached. Thank you for your help. Stagg Shelton. On Aug 23, 2008, at 10:08 AM, Iñaki Baz Castillo wrote:

The auto_aliases default is "on" according to the core cookbook for 1.3.x so I did not add that to my configuration. I did add alias=8.17.32.184:5060 per your suggestion and the ACK still continues in the infinite loop until the max hops is reached. I am using loose_route() if (has_totag() == true). I am pasting a copy of my openser config below for additional information on my specific configuration. Thank You Stagg Shelton # # $Id: openser.cfg 3542 2008-01-16 15:00:34Z miconda $ # # OpenSER basic configuration script # by Anca Vamanu <anca(a)voice-system.ro> # # Please refer to the Core CookBook at http://www.openser.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # ####### Global Parameters ######### debug=3 log_stderror=no log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ disable_tcp=yes port=5060 alias=8.17.32.184:5060 ####### Modules Section ######## #set module path mpath="//lib/openser/modules/" loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" loadmodule "uri_db.so" loadmodule "uri.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "avpops.so" loadmodule "alias_db.so" loadmodule "permissions.so" loadmodule "carrierroute.so" loadmodule "snmpstats.so" loadmodule "dialog.so" loadmodule "group.so" loadmodule "options.so" /* uncomment next line for multi-domain support NOTE: a DB (like 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" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) modparam("rr", "append_fromtag", 0) # ----- 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", "") # ----- 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", "db_flag", 1) modparam("acc", "db_missed_flag", 2) #modparam("acc","db_extra", "acc_extra=$var(accounting); acc_to= $var(numdialed)") modparam("acc","db_extra", "acc_extra=$var(accounting); acc_to= $var(numdialed); acc_direction=$var(direction)") #modparam("acc","db_extra", "acc_extra=$au; email=$avp(s:email)") modparam("acc", "db_url", "mysql://openser:oserdbpass@localhost/ openser") # ----- usrloc params ----- modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url","mysql://openser:oserdbpass@localhost/ openser") # ----- auth_db params ----- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url","mysql://openser:oserdbpass@localhost/ openser") modparam("auth_db", "load_credentials", "") # ----- alias_db params ----- modparam("alias_db", "db_url","mysql://openser:oserdbpass@localhost/ openser") # ----- permissions params ----- modparam("permissions", "db_url","mysql://openser:oserdbpass@localhost/ openser") # ----- carrierroute params ----- modparam("carrierroute", "config_source", "db") modparam("carrierroute", "db_url", "mysql:// openser:oserdbpass@localhost/openser") # ----- smpstats params ----- modparam("snmpstats", "sipEntityType", "registrarServer") modparam("snmpstats", "sipEntityType", "proxyServer") modparam("snmpstats", "snmpCommunity", "snmppass") modparam("snmpstats", "snmpgetPath", "/usr/bin/") # ----- dialog params ----- modparam("dialog", "enable_stats", 1) modparam("dialog", "dlg_flag", 4) modparam("dialog", "db_mode", 1) modparam("dialog", "db_url", "mysql://openser:oserdbpass@localhost/ openser") # ----- group params ----- modparam("group", "db_url", "mysql://openser:oserdbpass@localhost/ openser") ####### Routing Logic ######## # main request routing logic route{ $var(accounting) = ""; $var(numdialed) = ""; $var(direction) = ""; if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } # Options responder if (uri==myself) { if ((method==OPTIONS) && (! uri=~"sip:.*[@]+.*")) { options_reply(); } } if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accouting ... setflag(3); # ... even if the transaction fails } route(1); } else { if ( is_method("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 t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) setflag(1); setflag(2); t_relay(); exit; } t_check_trans(); if (!(method=="REGISTER")) { if (!allow_trusted()) { if (!proxy_authorize("", "subscriber")) { $var(debug) = proxy_authorize("", "subscriber"); xlog("L_INFO", "Proxy authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); xlog("Not Proxy Authorize: $var(debug)"); proxy_challenge("", "0"); exit; } consume_credentials(); # caller authenticated } } # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); if (method=="INVITE"||method=="BYE"||method=="CANCEL") { setflag(4); } if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); } # requests for my domain if (is_method("PUBLISH")) { sl_send_reply("503", "Service Unavailable"); exit; } if (is_method("REGISTER")) { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if (!check_to()) { sl_send_reply("403","Forbidden auth ID"); exit; } if (!save("location")) sl_reply_error(); exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } alias_db_lookup("dbaliases"); $var(accounting) = $rU; xlog ("After alias db lookup: $rU"); remove_hf("Diversion"); if (!lookup("location")) { switch ($retcode) { case -1: case -3: if (uri=~"^sip:[0-9]{10}@") # Domestic PSTN { route(4); exit; } else if (uri=~"^sip:1[0-9]{10}@") # Domestic PSTN { route(4); exit; } else if (uri=~"^sip:011[0-9][0-9]+@") # International PSTN { route(3); exit; }; t_reply("404", "User Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } else { $rU = $oU; $var(numdialed) = $rU; $var(direction)="inbound"; } # when routing via usrloc, log the missed calls also setflag(2); route(1); } route[1] { # for INVITEs enable some additional helper routes if (is_method("INVITE")) { xlog("aubrey direction $var(direction)"); setflag(1); setflag(4); t_on_branch("2"); t_on_reply("2"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); }; exit; } route[3] { # ----------------------------------------------------------------- # PSTN International Handler # ----------------------------------------------------------------- $var(accounting) = $au; $var(numdialed) = $rU; $var(direction)="outbound"; if(!is_user_in("Credentials", "international")){ sl_send_reply("403", "Forbidden: No Intl Service"); exit; } cr_user_rewrite_uri("sip:$au@.*","mydomain.net"); $avp(s:inv_timeout) = 45; route(1); } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- $var(accounting) = $au; $var(numdialed) = $rU; $var(direction)="outbound"; cr_user_rewrite_uri("sip:$au@.*","mydomain.net"); $avp(s:inv_timeout) = 45; route(1); } branch_route[2] { xlog("new branch at $ru\n"); } onreply_route[2] { xlog("incoming reply $rm\n"); xlog("$rb\n"); } failure_route[1] { if (t_was_cancelled()) { exit; } } On Aug 25, 2008, at 5:04 AM, Iñaki Baz Castillo wrote:

Hi! AFAIS the client is buggy (or is there a NAT ALG/Firewall between client and SIP proxy?). Compare the Contact header in the 200 OK and the request URI in the ACK. They MUST be the same!!! regards klaus U +0.000315 8.17.32.184:5060 -> 63.209.207.135:5060 SIP/2.0 200 OK* Via: SIP/2.0/UDP 63.209.207.135:5060;branch=z9hG4bK-8921-48b022df-dcaa3e6a-2f5ec169* Record-Route: <sip:8.17.32.184;lr=on;did=952.4d684275>* From: Anonymous <sip:restricted@63.209.207.135>;tag=88cfd13f-13c4-48b022df-dcaa3e6a-b4657f0* To: <sip:+16783832765@8.17.32.184:5060>;tag=as40da5b97* Call-ID: ATLMGC0720080823144655027771(a)209.244.63.15* CSeq: 1 INVITE* User-Agent: Asterisk PBX* Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY* Supported: replaces* Contact: <sip:+16783832765@8.17.32.19>* Content-Type: application/sdp* Content-Length: 180* U +0.072541 63.209.207.135:5060 -> 8.17.32.184:5060 ACK sip:+16783832765@8.17.32.184 SIP/2.0* From: Anonymous <sip:restricted@63.209.207.135>;tag=88cfd13f-13c4-48b022df-dcaa3e6a-b4657f0* To: <sip:+16783832765@8.17.32.184:5060>;tag=as40da5b97* Call-ID: ATLMGC0720080823144655027771(a)209.244.63.15* CSeq: 1 ACK* Via: SIP/2.0/UDP 63.209.207.135:5060;branch=z9hG4bK-8922-48b022e5-dcaa5757-3884948f* Max-Forwards: 15* Contact: <sip:restricted@63.209.207.135:5060;transport=udp>* Route: <sip:8.17.32.184;lr;did=952.4d684275>* Content-Length: 0* Stagg Shelton schrieb:

This is exactly what I told the carrier engineer about the problem. The contact in the 200 OK is my Asterisk PBX the RURI in the ACK (returning from the carrier) is my OpenSER. I informed the carrier that according to my interpretation of the RFC 3261 the RURI in the ACK MUST match the contact in the 200 OK with SDP. The carrier interop engineer sent me the following remark after I notified them of the issue, and began pressing them on the way their ACK was created. == BEGIN == The Request URI field is used for routing of initial dialog messages. If state is not kept in [Open]SER by using the tm module, the SIP message can be routed based on the VIA or contact header, or worst case follow the same routing rules based on the original Request URIs, just like the original INVITE message did. == END == After some subsequent communication where I again questioned the way in which their system creates the ACK. I received the below from them. == BEGIN == All I was saying is that based on the way your system handles messages we send to it, your statement about changing the Request-URIs of ACKs based on values from the SDP of the 200 OK is not the case. I believe that some systems do however update the ACK’s Request-URI based on the Contact header field from the 200 response, but most system’s don’t route the ACK based on its Request-URI when keeping state. I have another question based on our use of SER internally. When we send the initial INVITE, the Request-URI is set as the number at (@) the address of your [Open]SER proxy. SER will then re-write the request URI of the INVITE based on the logic in the local ser.cfg file and any local location/alias database entries used in that logic and forward the message to the destination, in this case the PBX, while adding a Record-Route header with LR set to ensure SER stays in the dialog in terms of signaling as well as adding a VIA header with a branch tag used to mark the dialog for stateful processing (when the TM module is used). In the case of stateful processing, these values are used to maintain the same signaling path for subsequent messages within the same dialog. When stateless routing is used via the SL module, all new requests (including an ACK to a 200 response) should follow the same routing logic as the initial request. Therefore, when the ACK arrives at your SER proxy, the same routing logic should be applied to the ACK as was the original INVITE, and it should be forwarded on to the correct destination. If this is not the case, then either there must be some kind of state being tracked or the logic has been written to handle ACKs differently on purpose as this would be the only way that the handling and routing (especially the computation of the final request- uri) would be different for an ACK from an INVITE. == END == At this point I am ready to address any issue with my OpenSER configuration that can be identified, but if the problem is actually due to the ACK not being constructed correctly, I have to take off the technical hat, and put on the business hat, and try to get them to look at their systems, and push their vendors for support in this issue. Thank You Stagg Shelton On Aug 25, 2008, at 9:52 AM, Klaus Darilion wrote:

As Iñaki said .... and one more - the is no difference if the routing in the SIP proxy is done stateless or stateful. regards klaus Stagg Shelton wrote:

I agree, and this is the first provider that we have had any issues with. My production OpenSER is supplying service to my end users from multiple providers using, sonus, broadsoft, & nextone without any issue. I have not been able to identify from packet captures which software or product this carrier is using. They have eluded in some comments that they are using SER to some capacity, but I do not have any confirmation. Thanks Stagg Shelton On Aug 25, 2008, at 12:17 PM, Iñaki Baz Castillo wrote:

El Monday 25 August 2008 18:22:19 Stagg Shelton escribió: Stagg, the point here is that your provider says: "most system’s don’t route the ACK based on its Request-URI when keeping state." It doesn't matter what "others" do, the fact is that the provider CANNOT send a WRONG formed ACK by asumming that the client will handle it in a exotic way instead of the way RFC3261 defines (the only standard). So the excuse of your provider is basically: - "With other clients I have not this problem". - "Most of them accept my ACK since they don't handle it as RFC3261 defines". Your provider must recognice that him ACK is wrong, regardless of how you proxy handles it (in fact it handles it correctly). And he MUST fix it. Pain of ignorant carriers... -- Iñaki Baz Castillo ibc(a)in.ilimit.es

El Lunes, 25 de Agosto de 2008, Klaus Darilion escribió: One question more: If you initiate an outgoing call through that provider, is it correctly established? In that case try to call to a PSTN phone of you, and hang up the call from the PSTN in order to receive a BYE from the provider. Is that BYE correctly generated? it should have a "Route: sip:openser_ip" and a RURI like the Contact you send in the INVITE. -- Iñaki Baz Castillo

El Monday 25 August 2008 21:47:15 Stagg Shelton escribió: Yes, the BYE is correctly formed, so your provider has a problem when generating the ACK for a 200 OK. -- Iñaki Baz Castillo ibc(a)in.ilimit.es