sr-users October 2004

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[Serusers] SER requirements
by Wee Liat 26 Oct '04

26 Oct '04

Hi, I would like to know if Fedora Core 2 fits the requirements of SER 0.8.14? -- Regards, Wee Liat

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[Serusers] Multiple UAs behind a single NAT
by Allan 26 Oct '04

26 Oct '04

Hi, Can SER be configured to allow two or more UAs to be NATed behind the same IP address? Also, is it possible to enable SER to see that the UAs are located behind the same address and, after call setup, let the UAs speak directly to each other? Could the ser.cfg alone support these features or would other changes need to be made? Regards, Allan ______________________________________________________________________ Post your free ad now! http://personals.yahoo.ca

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[Serusers] (no subject)
by dev2003 26 Oct '04

26 Oct '04

serusers，您好！ how to improve KPhone poor vedio effect? kphone-vic can not get good vedio effect.how to do? 　　　　　　　　致礼！　　　　　　　　dev2003 　　　　　　　　dev2003(a)mail.ustc.edu.cn 　　　　　　　　　　2004-10-26

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[Serusers] SER + SEMS + isdngw problem
by Socrates Varakliotis 26 Oct '04

26 Oct '04

Hi All: I'm trying to make an outgoing call from Kphone to an ISDN line. Although the called phone rings, when I pickup the call drops and I get the following errors on the logfiles: Has anyone come across this? Any suggestions welcome. Regards, -- Socrates. ------------------- SER.log --------------------------- ... sip_request.isdngw INVITE 76793696 . 128.16.64.71 128.16.64.71 . sip:576793696@128.16.64.71 sip:root@128.16.64.71:5062;transport=udp "Socrates Varakliotis" <sip:6222@shaggy.cs.ucl.ac.uk> <sip:576793696@128.16.64.71> 183464563 64247849 . 5593 55871:1543819790 . . P-MsgFlags: 0 . v=0 o=username 0 0 IN IP4 128.16.64.71 s=The Funky Flow c=IN IP4 128.16.64.71 t=0 0 m=audio 32802 RTP/AVP 0 97 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 > 5(4251) DEBUG: write_to_vm_fifo: write completed 5(4251) DEBUG: add_to_tail_of_timer[0]: 0x402e35a4 5(4251) DEBUG:destroy_avp_list: destroing list (nil) 5(4251) receive_msg: cleaning up 9(4255) ERROR: fifo_server: command t_reply is not available 9(4255) ERROR: fifo_server: command must begin with :: isgndw: ringing 9(4255) ERROR: fifo_server: command must begin with :: 000010A77D8E0A09 9(4255) ERROR: fifo_server: command must have at least 3 chars 9(4255) INFO: fifo_server: command empty 9(4255) ERROR: fifo_server: command t_reply is not available 9(4255) ERROR: fifo_server: command must have at least 3 chars 9(4255) ERROR: fifo_server: command must begin with :: 000010A77D8E0A09 9(4255) ERROR: fifo_server: command must begin with :: Content-Type: application/sdp 9(4255) ERROR: fifo_server: command must begin with :: v=0 9(4255) ERROR: fifo_server: command must begin with :: s=session 9(4255) ERROR: fifo_server: command must begin with :: t=0 0 9(4255) ERROR: fifo_server: command must begin with :: a=rtpmap:0 PCMU/8000 9(4255) INFO: fifo_server: command empty 9(4255) ERROR: fifo_server: command t_reply is not available 9(4255) ERROR: fifo_server: command must begin with :: could not send response. 9(4255) ERROR: fifo_server: command must begin with :: 000010A77D8E0A09 9(4255) ERROR: fifo_server: command must have at least 3 chars 9(4255) INFO: fifo_server: command empty 10(4257) DEBUG: timer routine:0,tl=0x402e35a4 next=(nil) ------------------- SEMS.log --------------------------- .... 180 isgndw: ringing 55871:1543819790 000010A77D8E0A09 Contact: <sip:76793696@128.16.64.71> . . > (4263) DEBUG: write_to_fifo (SerClient.cpp:244): Write to fifo: completed (4263) DEBUG: getReturnCode (AmRequest.cpp:357): response from Ser: 500 command 't_reply' not available (4263) ERROR: reply (AmRequest.cpp:335): AmRequestUAS::reply: 500 command 't_reply' not available (4263) DEBUG: cb_connect_active_ind (IsdnCapiConnection.cpp:282): callback funtion called: cb_connect_active_ind! (4263) DEBUG: cb_connect_active_ind (IsdnCapiConnection.cpp:290): physical connection established, now creating logical connection (4263) DEBUG: cb_connect_b3_conf (IsdnCapiConnection.cpp:308): connected, now unlocking mutex (4263) DEBUG: changePSTNState (IsdnCall.cpp:348): CALL: PSTN state (outbound): from (CONNECTING|CONNECTING) to (CONNECTING|CONNECTED) (4263) DEBUG: changePSTNState (IsdnCall.cpp:550): DONE: PSTN state (outbound): from (CONNECTING|IDLE) to (CONNECTING|CONNECTING)

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[Serusers] BYE message to SER.
by Ricardo Martinez 25 Oct '04

25 Oct '04

Hello list. I have a problem-question regarding to the BYE message reaching SER. For the normail example i have a endpoint registered in SER and a PSTN-Gateway receiving the outgoing traffic. For this enviroment when i made a call from my endpoint to the PSTN gateway and the Gateway send the BYE (at the end of the call) this is what i see in the debug in SER. U gw1.mydomain.com:52154 -> sersip.mydomain.com:5060 BYE sip:005622408196@sersip.mydomain.com:5060 SIP/2.0 Via: SIP/2.0/UDP gw1.mydomain.com:5060 From: <sip:005622408196@sersip.mydomain.com>;tag=A0272B49-B2D To: <sip:5555832351@sersip.mydomain.com>;tag=ae4208f2a4 Date: Mon, 25 Oct 2004 21:34:02 GMT Call-ID: ae229c42-f2c3-08f3-81f2-0002a400f1e9(a)xx.xx.xx.42 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 6 Route: <sip:5555832351@xx.xx.xx.42:5060> Timestamp: 1098740047 CSeq: 101 BYE Content-Length: 0 # U sersip.mydomain.com:5060 -> xx.xx.xx.42:5060 BYE sip:5555832351@xx.xx.xx.42:5060 SIP/2.0 Record-Route: <sip:005622408196@sersip.mydomain.com;ftag=A0272B49-B2D;lr=on> Via: SIP/2.0/UDP sersip.mydomain.com;branch=z9hG4bK4c65.37369754.0 Via: SIP/2.0/UDP gw1.mydomain.com:5060 From: <sip:005622408196@64.76.148.231>;tag=A0272B49-B2D To: <sip:5555832351@sersip.mydomain.com>;tag=ae4208f2a4 Date: Mon, 25 Oct 2004 21:34:02 GMT Call-ID: ae229c42-f2c3-08f3-81f2-0002a400f1e9(a)xx.xx.xx.42 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 5 Timestamp: 1098740047 CSeq:101 BYE Content-Length: 0 # U xx.xx.xx.42:5060 -> sersip.mydomain.com:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP sersip.mydomain.com;branch=z9hG4bK4c65.37369754.0 Via: SIP/2.0/UDP gw1.mydomain.com:5060 From: <sip:005622408196@sersip.mydomain.com>;tag=A0272B49-B2D To: <sip:5555832351@sersip.mydomain.com>;tag=ae4208f2a4 Call-ID: ae229c42-f2c3-08f3-81f2-0002a400f1e9(a)xx.xx.xx.42 CSeq: 101 BYE User-Agent: AddPac SIP Gateway Content-Length: 0 Record-Route: <sip:005622408196@sersip.mydomain.com;ftag=A0272B49-B2D;lr=on> # U sersip.mydomain.com:5060 -> gw1.mydomain.com:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP gw1.mydomain.com:5060 From: <sip:005622408196@sersip.mydomain.com>;tag=A0272B49-B2D To: <sip:5555832351@sersip.mydomain.com>;tag=ae4208f2a4 Call-ID: ae229c42-f2c3-08f3-81f2-0002a400f1e9(a)xx.xx.xx.42 CSeq: 101 BYE User-Agent: AddPac SIP Gateway Content-Length: 0 Record-Route: <sip:005622408196@sersip.mydomain.com;ftag=A0272B49-B2D;lr=on> As you can see the call is ended succesfully and there is no problem. Now, instead of having a PSTN-Gateway i use a SIP-H323 translator and all the calls are going out through this device. When i made a call from my endpoint and the SIP-H323 send the BYE message (at the end of the call) the debug shows this : U sipquest.mydomain.com:5060 -> sersip.mydomain.com:5060 BYE sip:1112001003@sersip.mydomain.com;ftag=25421eefa4;lr=on SIP/2.0 Via: SIP/2.0/UDP sipquest.mydomain.com:5060; branch=z9hG4bKsKpiQfXRAwAYAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAB_ From: <sip:1112001003@sersip.mydomain.com>;tag=3896741524 To: <sip:5555832351@sersip.mydomain.com>;tag=25421eefa4 CSeq: 1 BYE Call-ID: 25229c42-16c9-1e95-81ef-0002a400f1e9(a)xx.xx.xx.42 Contact: <sip:1112001003@sipquest.mydomain.com> Route: <sip:5555832351@sersip.mydomain.com> Max-Forwards: 69 Content-Length: 0 # U sersip.mydomain.com:5060 -> sipquest.mydomain.com:5060 SIP/2.0 404 Not Found IT! Via: SIP/2.0/UDP sipquest.mydomain.com:5060; branch=z9hG4bKsKpiQfXRAwAYAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAB_;received=sipquest .mydomain.com From: <sip:1112001003@sersip.mydomain.com>;tag=3896741524 To: <sip:5555832351@sersip.mydomain.com>;tag=25421eefa4 CSeq: 1 BYE Call-ID: 25229c42-16c9-1e95-81ef-0002a400f1e9(a)xx.xx.xx.42 Server: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 Warning: 392 sersip.mydomain.com:5060 "Noisy feedback tells: pid=15285 req_src_ip=sipquest.mydomain.com req_src_port=5060in_uri=sip:1112001003@sersip.mydomain.com;ftag=25421eefa4;l r=on out_uri=sip:1112001003@sersip.mydomain.com;ftag=25421eefa4;lr=on via_cnt==1" My question is : Why the call is ended succesfully in the first case and in the second case SER answer with a "NOT FOUND"? I only have registered in my location table the endpoints. Here is my ser.cfg file. # # $Id: ser.cfg,v 1.21.2.2 2003/10/13 22:53:06 jiri Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=0 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) #Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes listen=xx.xx.148.231 alias=sersip.mydomain.com check_via=yes # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 #children=4 fifo="/tmp/ser_fifo" fifo_mode=0666 # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local//lib/ser/modules/mysql.so" loadmodule "/usr/local//lib/ser/modules/sl.so" loadmodule "/usr/local//lib/ser/modules/tm.so" loadmodule "/usr/local//lib/ser/modules/rr.so" loadmodule "/usr/local//lib/ser/modules/maxfwd.so" loadmodule "/usr/local//lib/ser/modules/usrloc.so" loadmodule "/usr/local//lib/ser/modules/registrar.so" loadmodule "/usr/local//lib/ser/modules/textops.so" loadmodule "/usr/local//lib/ser/modules/exec.so" # Para trabajar con NAT loadmodule "/usr/local//lib/ser/modules/mediaproxy.so" loadmodule "/usr/local//lib/ser/modules/domain.so" loadmodule "/usr/local//lib/ser/modules/dbtext.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" loadmodule "/usr/local/lib/ser/modules/group_radius.so" loadmodule "/usr/local/lib/ser/modules/acc.so" # ----------------- setting module-specific parameters --------------- # ----------------- setting module-specific parameters --------------- # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # module authorization modparam("auth_radius","radius_config","/usr/local/etc/radiusclient/radiuscl ient.conf") modparam("auth_radius","service_type",15) #module accounting modparam("acc","radius_config","/usr/local/etc/radiusclient/radiusclient.con f") modparam("acc","log_level",1) modparam("acc", "service_type", 15) modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 3) modparam("acc", "failed_transactions", 1) modparam("acc", "report_ack", 0) #group radius modparam("group_radius","radius_config","/usr/local/etc/radiusclient/radiusc lient.conf") modparam("group_radius", "use_domain", 1) #module mediaproxy modparam("mediaproxy", "natping_interval", 60) modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") #modparam("mediaproxy", "sip_asymmetrics", "/usr/local/etc/ser/sip-asymmetrics-clients") #modparam("mediaproxy", "rtp_asymmetrics", "/usr/local/etc/ser/rtp-asymmetrics-clients") modparam("registrar", "nat_flag", 5) #module domain modparam("domain", "db_url", "/usr/local/etc/ser/domaintables") modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") # ------------------------- request routing logic ------------------- # main routing logic route { # Llamada a Rutina Externa para cambio de numero # if (method=="INVITE") { # exec_dset("/usr/local/etc/ser/redirect.sh"); # }; # 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"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # subsequent messages withing a dialog should take the # path determined by record-routing record_route(); if (loose_route()) { if (method=="BYE" || method=="CANCEL") { log(1, "NAT: BYE o CANCEL recibido --> terminando la sesion de media\n"); end_media_session(); setflag(1); }; t_relay(); break; }; # Maracamos las llamadas para accounting setflag(1); if (client_nat_test("1")) { log(1, "NAT: Requerimiento de IP privada --> fixed contact (en rutina principal)\n"); setflag(5); force_rport(); fix_contact(); append_hf("P-hint: fixed NAT contact for request\r\n"); }; # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Challenge/Response if ( !radius_www_authorize("")) { www_challenge("", "1"); break; }; # Mandamos un mensaje de Log si es que un cliente NATEADO se registro. if (isflagset(5)) { log(1, "NAT: Cliente NAT'eado registrado (dentro del chequeo de REGISTER)\n"); }; save("location"); break; }; # FIN DE CHEQUEO DE REGISTRO # lookup("aliases"); # CHEQUEO EL INVITE DE LA LLAMADA if (method=="INVITE" && search("^From: <sip:.*@xx.xx.148.231>")) { if ( !radius_proxy_authorize("")) { proxy_challenge("", "1"); log (1, "LOG: Llamada no AUTORIZADA\n"); break; }; }; if ( (uri=~"^sip:0.*@.*") ){ if (method=="INVITE") { log(1,"FORWARDING : Llamada hacia plataforma H.323\n"); route(3); break; }; }; # Llamadas hacia dominio H.323 if ( (uri=~"^sip:111.*@.*") | (uri=~"^sip:777.*@.*") | (uri=~"^sip:333.*@.*") ){ if (method=="INVITE") { log(1,"FORWARDING : Llamada hacia plataforma H.323\n"); route(1); break; }; }; # Llamadas SIP locales if (!lookup("location")) { #Usuarios que no estan en "location" database. route(4); break; }; setflag(3); # Manejo para llamadas NAT'das if (method=="INVITE") { log(1, "ROUTING: Reply processing (online user) enabled to handle NAT'd resonses\n"); t_on_reply("1"); if ( (isflagset(5)) && (method=="INVITE" || method=="ACK") ) { log(1, "NAT: Invite received --> enabling media proxy (else del lookup 2do if)\n"); use_media_proxy(); append_hf("P-hint: request forced to media proxy\r\n"); }; }; # Do it ? (Ricardo Martinez) }; append_hf("P-hint: USRLOC\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } /* end of initial routing logic */ #--------- Route 1 hacia SIPQUEST ------------------------------- route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP # Esta NATEADO? if (isflagset(5)) { log(1, "NAT: At least one participant NAT'd --> enable reply processing (hacia GW)\n"); t_on_reply("1"); if (method=="INVITE") { log(1, "NAT: Invite received --> enabling proxied media session (hacia GW)\n"); setflag(1); use_media_proxy(); }; }; rewritehostport("sipquest.mydomain.com:5060"); append_hf("P-hint: GATEWAY\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } #------- Route 3 : Trafico de Internet hacia la PSTN ------------------- route[3] { setflag(1); #Esta NETEADO? if (isflagset(5)) { log(1, "NAT: At least one participant NAT'd --> enable reply processing (hacia GW)\n"); t_on_reply("1"); if (method=="INVITE") { log(1, "NAT: Invite received --> enabling proxied media session (hacia GW)\n"); #setflag(1); use_media_proxy(); }; }; rewritehostport("gw1.mydomain.com:5060"); append_hf("P-hint: GATEWAY\r\n"); t_relay(); break; } #------ Route 4 : Llamadas a usuarios Off-line ------------------------ route[4] { if (!t_newtran()) { sl_reply_error(); }; if (!t_reply("404", "Not Found IT!")) { sl_reply_error(); }; break; } # ---------------------------- Begin On-Reply Routes -------------------------- onreply_route[1] { # If we've got here, it's because we've previously passed through a block # which handles NAT requests and has set a t_on_reply condition. DB 03-08-2004 if (status =~ "(183)|(2[0-9][0-9])") { if (client_nat_test("1") || isflagset(5)) { log(1, "NAT: Reply from NAT'd client --> fixing contact (onreply_route)\n"); fix_contact(); use_media_proxy(); setflag(5); }; log(1, "NAT: NAT'd transaction answered --> enabling media proxy (onreply_route)\n"); # use_media_proxy(); }; } # ------------------------------ End On-Reply Routes -------------------------- Can someone please help me?. Ricardo Martinez

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[Serusers] rtpproxy+pstn call
by Trung Nguyen 25 Oct '04

25 Oct '04

Hi Serusers, I have been playing around this logic for some time now and it's giving me a bit of headache. Any help will be very appreciated. Here's what I got. I use the default setting for rtpproxy and tested all the nat scenarios from PC-PC work great. Now I want to have the PC to call out to PSTN and only allow registered client to be able to make call to pstn gateway and at the same time to go thru rtpproxy. My gateway ip address is 64.200.219.134 port 5060, it's a cisco as5350. Below is my configuration, where should I put the setting and any hint or example will be very appreciated. ---------------------- # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=7 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ listen=64.200.219.135 listen=127.0.0.1 alias=unlimitedtalk.net alias=64.200.219.135 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/acc.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # ------------- tm parameters modparam("tm", "fr_timer", 12) modparam("tm", "fr_inv_timer", 24) # ------------- rr parameters # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------- accounting parameters modparam("acc", "log_missed_flag", 3) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) # ------------- usrloc parameters # 2 enables write-back to persistent mysql storage for speed # disable=0, write-through=1 modparam("usrloc", "db_mode", 2) # minimize write back window - default is 60 seconds modparam("usrloc", "timer_interval", 10) # database location modparam("usrloc", "db_url", "mysql://ser:heslo@localhost/ser") # ------------- auth parameters # database location modparam("auth_db", "db_url", "mysql://ser:heslo@localhost/ser") # allows clear text passwords in the mysql database modparam("auth_db", "calculate_ha1", yes) # name of password column in mysql database modparam("auth_db", "password_column", "password") # !! Nathelper modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT # ----------------- setting module-specific parameters --------------- # ------------------------- 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"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # 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 record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # !! Nathelper # Special handling for NATed clients; first, NAT test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding is used); also, # the received test should, if completed, should check all # vias for rpesence of received if (nat_uac_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart enough to be symmetric. In some phones it takes a configuration # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is # called "symmetric media" and "symmetric signalling". fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; # 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"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # 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") { # Uncomment this if you want to use digest authentication if (!www_authorize("unlimitedtalk.net", "subscriber")) { www_challenge("unlimitedtalk.net", "0"); break; }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # !! Nathelper if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } # !! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); }; } ---------------------------- Thanks in advance, Ted _______________________________ Do you Yahoo!? Declare Yourself - Register online to vote today! http://vote.yahoo.com

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[Serusers] serweb installation trouble
by rosita sita 25 Oct '04

25 Oct '04

hii, i have problems in my serweb. when my clients tried to subscribe,they are successfull in serweb but there is no message that my client received in their e-mail. besides that,i checked, my clients are in pending tables in mysql database so they can't login. anybody can help me????please your advices.i'm newbie in making SER. regards, sita -- Sita Rosita ___________________________________________________ Take your business online with Officemaster. Sign up for a free trial today! http://www.officemaster.net

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[Serusers] Forwarding to pstn, accounting problem
by Pavel Siderov 25 Oct '04

25 Oct '04

After I made my accounting up and working I've changed something and it's not working now would you help me please. The problem is that when I forward the call to pstn gateway when user closes the line the "BYE" request is not written in the database - it exists but it's not written - here it is 3(17265) SIP Request: 3(17265) method: <BYE> 3(17265) uri: <sip:pavel@192.168.2.111:19454> 3(17265) version: <SIP/2.0> 3(17265) parse_headers: flags=1 3(17265) get_hdr_field: cseq <CSeq>: <1> <BYE> Here is part of my ser.cfg modparam("acc", "log_flag", 1) modparam("acc", "log_fmt", "cdfimorstup") modparam("acc", "db_url", "mysql://user:pass@192.168.2.113:3306/ser") modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 1) modparam("acc", "log_missed_flag", 1) #modparam("acc", "report_cancels", 1) modparam("acc", "report_ack", 1) ....... if (uri=~"^sip:[0-9]*@sip.mydomain.com") { if (is_user_in("From", "pstn")){ rewritehost("pstn_gw_ip"); append_branch("pstn_gw_ip"); setflag(1); record_route(); force_rtp_proxy(); t_relay(); t_on_reply("1"); if (loose_route()) { t_relay(); break; }; break; } else { sl_send_reply("399", "Your account balance is low!"); }; break; }; What's is the problem please help me. Thanks in advance! Pavel Siderov

2 2

[Serusers] Problem with SER-Mediaproxy and VPN
by Bernie Hoeneisen 25 Oct '04

25 Oct '04

Hi! I have the following setup: * Client A is behind a VPN, which make a NAT as follows: - IP Address on the interface: 80.32.110.203 - IP address for VPN: 80.32.23.17 (as seen outside the VPN-NAT) - The client beliefs that it has IP 80.32.110.203, and uses this for its SIP and SDP messages. * Client B uses IP 80.32.6.184 (no VPN) * The SER-Mediaproxy runs on 80.32.10.95 If Client A (VPN) calls client B, there is no problem: SER and SERMediaproxy handle it correctly. So far so good. But, if Client B wants to call Client A (the one with VPN), no audio is transmitted. While tracing, I figured out the following: - Client A sends its RTPs from 80.32.23.17 to the SER-Mediaproxy, which the SER-Mediaproxy considers a as 3rd party, and therefore it ignors RTP packets coming from that source IP. - Client B sends its RTP to the SER-Mediaproxy, which forwards them to 80.32.110.203 (the Interface, which Client A has put to the SDP). Those are ignored by the VPN on the host of client A, as it only accepts packets belonging to the VPN tunel. --> Ergo, no RTP goes through. :-( Any ideas, on how I can tell SER-Mediaproxy to accept packets from 80.32.23.17 ? According to the logs the SER-Mediaproxy knows about this IP... cheers, Bernie PS: The mediaproxy.log look as follows: normaluser(a)test-sip.ch calls vpnuser(a)test-sip.ch ---------------------------------------------------- command request 000bfd32-e66e0027-0b258da4-1e55a8b9(a)80.32.6.184 80.32.6.184:16672:audio 80.32.6.184 test-sip.ch local 217.162.217.186 remote CSCO/7 info=from:normaluser@test-sip.ch,to:vpnuser@test-sip.ch,fromtag:000bfd32e66e00627e7b69c3-6f480ec5,totag:,dispatcher session 000bfd32-e66e0027-0b258da4-1e55a8b9(a)80.32.6.184: started. listening on 80.32.10.95:16604 command execution time: 2.80 ms command lookup 000bfd32-e66e0027-0b258da4-1e55a8b9(a)80.32.6.184 80.32.110.203:16384:audio 80.32.23.17 test-sip.ch local test-sip.ch unknown SJLabs-SJphone/1.30.235a info=from:normaluser@test-sip.ch,to:vpnuser@test-sip.ch,fromtag:000bfd32e66e00627e7b69c3-6f480ec5,totag:3505347165,dispatcher command execution time: 0.52 ms warning: Received packet from a third party: 80.32.23.17:16384 ^^^^^^^^^^^^^^^^^ session 000bfd32-e66e0027-0b258da4-1e55a8b9(a)80.32.6.184: caller signed in from 80.32.6.184:16672 (RTP) (will return to 80.32.6.184:16672) command lookup 000bfd32-e66e0027-0b258da4-1e55a8b9(a)80.32.6.184 80.32.110.203:16384:audio 80.32.23.17 test-sip.ch local test-sip.ch unknown SJLabs-SJphone/1.30.235a info=from:normaluser@test-sip.ch,to:vpnuser@test-sip.ch,fromtag:000bfd32e66e00627e7b69c3-6f480ec5,totag:3505347165,dispatcher command execution time: 0.20 ms command lookup 000bfd32-e66e0027-0b258da4-1e55a8b9(a)80.32.6.184 80.32.110.203:16384:audio 80.32.23.17 test-sip.ch local test-sip.ch unknown SJLabs-SJphone/1.30.235a info=from:normaluser@test-sip.ch,to:vpnuser@test-sip.ch,fromtag:000bfd32e66e00627e7b69c3-6f480ec5,totag:3505347165,dispatcher command execution time: 0.25 ms command lookup 000bfd32-e66e0027-0b258da4-1e55a8b9(a)80.32.6.184 80.32.110.203:16384:audio 80.32.23.17 test-sip.ch local test-sip.ch unknown SJLabs-SJphone/1.30.235a info=from:normaluser@test-sip.ch,to:vpnuser@test-sip.ch,fromtag:000bfd32e66e00627e7b69c3-6f480ec5,totag:3505347165,dispatcher command execution time: 0.21 ms session 7FB39E34-1DD2-11B2-A6E5-C208B29D816F(a)80.32.110.203: 0/0/0 packets, 0/0/0 bytes (caller/called/relayed) session 7FB39E34-1DD2-11B2-A6E5-C208B29D816F(a)80.32.110.203: ended (did timeout). command delete 000bfd32-e66e0027-0b258da4-1e55a8b9(a)80.32.6.184 info=dispatcher session 000bfd32-e66e0027-0b258da4-1e55a8b9(a)80.32.6.184: 515/0/515 packets, 103000/0/103000 bytes (caller/called/relayed) session 000bfd32-e66e0027-0b258da4-1e55a8b9(a)80.32.6.184: ended. command execution time: 0.47 ms vpnuser(a)test-sip.ch calls normaluser(a)test-sip.ch --------------------------------------------------- command request ADFBCDAC-1DD1-11B2-8626-975F6F226C9C(a)80.32.110.203 80.32.110.203:16386:audio 80.32.23.17 test-sip.ch local 80.32.6.184 remote SJLabs-SJphone/1.30.235a info=from:vpnuser@test-sip.ch,to:normaluser@test-sip.ch,fromtag:3505381983,totag:,dispatcher session ADFBCDAC-1DD1-11B2-8626-975F6F226C9C(a)80.32.110.203: started. listening on 80.32.10.95:16606 command execution time: 0.97 ms command lookup ADFBCDAC-1DD1-11B2-8626-975F6F226C9C(a)80.32.110.203 80.32.6.184:16674:audio 80.32.6.184 test-sip.ch local test-sip.ch unknown CSCO/7 info=from:vpnuser@test-sip.ch,to:normaluser@test-sip.ch,fromtag:3505381983,totag:000bfd32e66e00630d9f84f8-56b109d0,dispatcher command execution time: 0.50 ms session ADFBCDAC-1DD1-11B2-8626-975F6F226C9C(a)80.32.110.203: caller signed in from 80.32.23.17:16386 (RTP) (will return to 80.32.23.17:16386) session ADFBCDAC-1DD1-11B2-8626-975F6F226C9C(a)80.32.110.203: called signed in from 80.32.6.184:16674 (RTP) (will return to 80.32.6.184:16674) command delete ADFBCDAC-1DD1-11B2-8626-975F6F226C9C(a)80.32.110.203 info=dispatcher session ADFBCDAC-1DD1-11B2-8626-975F6F226C9C(a)80.32.110.203: 327/316/643 packets, 65400/63200/128600 bytes (caller/called/relayed) session ADFBCDAC-1DD1-11B2-8626-975F6F226C9C(a)80.32.110.203: ended. command execution time: 0.48 ms

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[Serusers] Max Forward Problem.
by Ricardo Martinez 25 Oct '04

25 Oct '04

Hello. I have been experimenting a weird problem in SER. I'm using a SIP-H323 protocol translator from SipQuest. All my callf from SER are going to the SIP-H323. When i made a call and the remote user (in SipQuest) sends a BYE message to SER i get the following answer. U sipquest.mydomain.com:5060 -> sersip.mydomain.com:5060 BYE sip:1111234567@sersip.mydomain.com;ftag=60425286a4;lr=on SIP/2.0 Via: SIP/2.0/UDP sipquest.magenta.cl:5060; branch=z9hG4bKsKpiQfXRAwAFAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA_ From: <sip:1111234567@sersip.mydomain.com>;tag=3896741524 To: <sip:5555832351@sersip.mydomain.com>;tag=60425286a4 CSeq: 1 BYE Call-ID: 60ce9b42-4559-5298-8186-0002a400f1e9(a)64.76.148.242 Contact: <sip:1111234567@sipquest.magenta.cl> Route: <sip:5555832351@sersip.mydomain.com> Max-Forwards: 69 Content-Length: 0 # U sipquest.mydomain.com:5060 -> sersip.mydomain.com:5060 BYE sip:1111234567@sersip.mydomain.com;ftag=60425286a4;lr=on SIP/2.0 Via: SIP/2.0/UDP sipquest.magenta.cl:5060; branch=z9hG4bKsKpiQfXRAwAFAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA_ From: <sip:1111234567@sersip.mydomain.com>;tag=3896741524 To: <sip:5555832351@sersip.mydomain.com>;tag=60425286a4 CSeq: 1 BYE Call-ID: 60ce9b42-4559-5298-8186-0002a400f1e9(a)64.76.148.242 Contact: <sip:1111234567@sipquest.magenta.cl> Route: <sip:5555832351@sersip.mydomain.com> Max-Forwards: 69 Content-Length: 0 # U sersip.mydomain.com:5060 -> sipquest.mydomain.com:5060 SIP/2.0 483 Too Many Hops Via: SIP/2.0/UDP sipquest.magenta.cl:5060;received=sipquest.mydomain.com; branch=z9hG4bKsKpiQfXRAwAFAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA_ From: <sip:1111234567@sersip.mydomain.com>;tag=3896741524 To: <sip:5555832351@sersip.mydomain.com>;tag=60425286a4 CSeq: 1 BYE Call-ID: 60ce9b42-4559-5298-8186-0002a400f1e9(a)64.76.148.242 Server: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 Warning: 392 sersip.mydomain.com:5060 "Noisy feedback tells: pid=14202 req_src_ip=sersip.mydomain.com req_src_port=5060in_uri=sip:1111234567@sersip.mydomain.com;ftag=60425286a4;l r=on out_uri=sip:1111234567@sersip.mydomain.com;ftag=60425286a4;lr=on via_cnt==70" But when the SER user sent yhe BYE message the call is procesed OK. Here is the BYE 64.76.148.242:5060 -> sersip.mydomain.com:5060 BYE sip:1111234567@sipquest.magenta.cl SIP/2.0 Via: SIP/2.0/UDP 64.76.148.242:5060;branch=z9hG4bK2c423483a4283 From: <sip:5555832351@sersip.mydomain.com>;tag=2c423483a4 To: <sip:1111234567@sersip.mydomain.com>;tag=3896741524 Call-ID: 2cce9b42-1f88-34fd-8183-0002a400f1e9(a)64.76.148.242 CSeq: 283 BYE Route: <sip:1111234567@sersip.mydomain.com;ftag=2c423483a4;lr=on> Date: Tue, 31 May 2005 02:39:19 GMT User-Agent: AddPac SIP Gateway Contact: <sip:5555832351@64.76.148.242> Content-Length: 0 Max-Forwards: 70 I'm using RADIUS and this problem is generating at least a 50 radius stop messages. I can send you the SER debug. I hope that someone can help me. Ricardo.-

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sr-users October 2004