sr-users

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4 participants
37456 discussions

[Serusers] voip peering
by Mindaugas Kuprys 19 Jun '06

19 Jun '06

Hi, Is there anyone running ser for peering with other voip providers using IPERX, Neustar and Arbinet provided services? I'm thinking to do that or should better use other solution like enum server http://www.ag-projects.com/ENUM.html? Regards, Mindaugas

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[Users] Sql possiblities of 1.0.1
by Fatih Cerit 19 Jun '06

19 Jun '06

Hi all Is it possible to do these with 1.0.1 SQL modules?? 1- is it possible to only insert the location queries instead of insert and update? 2- is it possible to lookup() from diffrent table (like a view) and save() location table? if possible which will queried from init_db. 3- is it possible to call function-select instead of a insert like select * from function_location('','','','','',''); thanks cerit

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[Users] planet VIP 152 T
by Joao Pereira 19 Jun '06

19 Jun '06

Hello to all Im testing a Planet 152 T phone and Im having some problems. Can someone tell me if this phone does URI dialing? And does it work behind NAT (does it need any special configuration on the SIP server)? Thanks Joao Pereira

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[Users] sip contact relay/forward/rewrite?
by Paul Smith 19 Jun '06

19 Jun '06

Hi, I am new to openser and trying to use it as a proxy i.e. I have a phone behind NAT, and another SIP server with all our account/registrar/routing details, I want to try mediaproxy as a far-end nat traversal solution to allow us to install phones without doing much (or anything) to the NAT firewalls.... The idea is that I set up phones to use Openser as Outbound proxy, and use symmetric RTP, and tell openser to nathelper to do regular keep alive pings. Then the phone should open SIP port to openser when registering, and OpenSER forwards the registration to the current sip server. Later invites from the current server should go back through Openser to the phone over the still open ip:port pair. MediaProxy on the openser should handle the RTP traffic between phone and old sip server. I have got ser and mediaproxy running and outgoing calls are fine. The problem is that incoming calls are not going through the proxy... the phone registers via the openser proxy, but when initiating an incoming call to the phone the registrar server sends the SIP invite straight to the NAT firewall, which rejects it... What do I need to do so that openSER tells the registrar to route SIP INVITES back through the openser server? I realise that at present I am looking at a very insecure open relay - I'll add more auth and domain checking when I get the basic outbound relay / media proxy bit working. PHONE ---- NAT ---- SER ----- SIP registrar Server with user details etc. Thanks.

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[Users] Can't forward to PSTN gateway
by Kenny Chua 19 Jun '06

19 Jun '06

Hi all, I'm having trouble allowing calls to my pstn gateway. I get some errors. These are the errors I get: I get the 404 Not Found error Here is the information I got from ngrep: U 2006/06/17 02:58:31.372496 68.xxx.xxx.17:5060 -> 192.168.1.20:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 192.168.1.1:5060;rport=5060;branch=z9hG4bK082837F1736B4E11832482B6D09012F5..From: K enny <sip:201@mydomain.myftp.org>;tag=2300590181..To: <sip:13373038435@mydomain.myftp.org>..Call-ID: 4B570DFD-0E2E-4F32-8B4B-05ADC419D0E7(a)192.168.1.20.. CSeq: 60451 INVITE..Server: OpenSer (1.0.1 (i386/linux))..Content-Length: 0..Warning: 392 192.168.1.5:5060 "Noisy feedback tells: pid=2874 req_src_ip=1 92.168.1.1 req_src_port=5060 in_uri=sip:13373038435@mydomain.myftp.org out_uri=sip:13373038435@pstn.gateway.net:5060 via_cnt==1".... U 2006/06/17 02:58:31.487474 192.168.1.5:5060 -> 65.xxx.xxx.xxx:5060 INVITE sip:13373038435@pstn.gateway.net:5060 SIP/2.0..Record-Route: <sip:192.168.1.5;ftag=2300590181;lr=on>..Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bK 7217.285e6331.0..Via: SIP/2.0/UDP 192.168.1.1:5060;rport=5060;branch=z9hG4bK082837F1736B4E11832482B6D09012F5..From: Kenny <sip:201@mydomain.myftp.org>;t ag=2300590181..To: <sip:13373038435@mydomain.myftp.org>..Contact: <sip:201@192.168.1.1:5060>..Call-ID: 4B570DFD-0E2E-4F32-8B4B-05ADC419D0E7(a)192.168.1.20 ..CSeq: 60451 INVITE..Max-Forwards: 69..Content-Type: application/sdp..User-Agent: X-Lite release 1105x..Content-Length: 304....v=0..o=201 10653130 1065 3152 IN IP4 68.226.254.17..s=X-Lite..c=IN IP4 192.168.1.1..t=0 0..m=audio 8000 RTP/AVP 0 8 3 98 97 101..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=rt pmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. U 2006/06/17 02:58:31.545023 65.200.189.132:5060 -> 192.168.1.5:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 68.226.254.17:5060;branch=z9hG4bK7217.285e6331.0..Via: SIP/2.0/UDP 192.168.1.1:5060;rport=5060;branch=z9hG4bK082837 F1736B4E11832482B6D09012F5..From: Kenny <sip:201@mydomain.myftp.org>;tag=2300590181..To: <sip:13373038435@mydomain.myftp.org>..Call-ID: 4B570DFD-0E2E-4F 32-8B4B-05ADC419D0E7@192.168.1.20..CSeq: 60451 INVITE..Content-Length: 0.... U 2006/06/17 02:58:31.661353 65.200.189.132:5060 -> 192.168.1.5:5060 SIP/2.0 500 Server Internal Error..To: <sip:13373038435@mydomain.myftp.org>;tag=3359519831-260907..From: Kenny <sip:201@mydomain.myftp.org>;tag=23005901 81..Call-ID: 4B570DFD-0E2E-4F32-8B4B-05ADC419D0E7@192.168.1.20..CSeq: 60451 INVITE..Contact: sip:13373038435@65.xxx.xxx.xxx:5060..Via: SIP/2.0/UDP 68.xx x.xxx.17:5060;branch=z9hG4bK7217.285e6331.0..Via: SIP/2.0/UDP 192.168.1.1:5060;rport=5060;branch=z9hG4bK082837F1736B4E11832482B6D09012F5..Content-Length : 0.... U 2006/06/17 02:58:31.661842 192.168.1.5:5060 -> xx.xxx.xxx.132:5060 ACK sip:13373038435@pstn.gateway.net:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bK7217.285e6331.0..From: Kenny <sip:201@mydomain.myftp.org>; tag=2300590181..Call-ID: 4B570DFD-0E2E-4F32-8B4B-05ADC419D0E7@192.168.1.20..To: <sip:13373038435@mydomain.myftp.org>;tag=3359519831-260907..CSeq: 60451 ACK..User-Agent: OpenSer (1.0.1 (i386/linux))..Content-Length: 0.... Below is my openser.cfg # # $Id: openser.cfg,v 1.5 2005/10/28 19:45:33 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ # put your server IP address here listen=192.168.1.5 alias=mydomain.myftp.org check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" modparam("auth_db|permissions|uri_db|usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); exit; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { route(1); }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (method=="REGISTER") { xlog("Registering\n"); route(2); return; }; lookup("aliases"); if(method =="INVITE"){ if( uri=~"^sip:1[0-9]*@"){ xlog("International call\n"); route(4); exit; }; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (!t_relay()) { sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!www_authorize("mydomain.myftp.org","subscriber")) { www_challenge("mydomain.myftp.org","0"); exit; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); return; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; xlog("Registered Successfully\n"); } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); lookup("aliases"); if (uri!=myself) { route(1); }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; route(1); } route[4]{ xlog("Before forward to host\n"); rewritehostport("pstn.gateway.net:5060"); xlog("After forward to host\n"); route(1); } What am I doin wrong? Why am I not able to dial a PSTN line? My PSTN gateway allows international calls. Any help is appreciated. Thank you. --------------------------------- Ring'em or ping'em. Make PC-to-phone calls as low as 1¢/min with Yahoo! Messenger with Voice.

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[Users] lawful legal intercept
by tele 19 Jun '06

19 Jun '06

hi, all is it possible to implement with an openser a legal interception? maybe with t_replicate() to another server. what about the media forking of the call? thanks :tele

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[Users] SIP Clients Can't REGISTER
by Hamid Ali Asgari 19 Jun '06

19 Jun '06

Hi, I have followed the Intrsuctions on setting up FreeRADIUS with OpenSER. (http://openser.org/docs/openser-radius-1.0.x.html) But the server doesn't allow the registration: Config: ============================================================================ ==================== # $Id$ # # radius config script # # ----------- global configuration parameters ------------------------ debug=7 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 listen=udp:192.168.10.18 alias="openser.org" alias="mydomain.com" #fifo="/tmp/openser_fifo" # ------------------ module loading ---------------------------------- mpath="/usr/local/lib/openser/modules/" loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "avpops.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "xlog.so" loadmodule "uri.so" loadmodule "acc.so" loadmodule "auth.so" loadmodule "auth_radius.so" loadmodule "group_radius.so" loadmodule "avp_radius.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc","db_url","mysql://openser:openserrw@localhost/openser") modparam("usrloc", "db_mode", 2) # -- acc params -- modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 1) modparam("acc", "service_type", 15) modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp") modparam("acc|auth_radius|group_radius|avp_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") # -- group_radius params -- modparam("group_radius", "use_domain", 1) # -- avpops params -- modparam("avpops", "avp_aliases", "day=i:101;time=i:102") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- 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; }; # check if user is suspended if(is_method("REGISTER|INVITE|MESSAGE|OPTIONS|SUBSCRIBE")) { if (radius_is_user_in("From", "suspended")) { sl_send_reply("403", "Forbidden - suspended"); exit; }; }; # 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); }; if(is_method("INVITE") && !has_totag()) { # set the acc flags setflag(1); setflag(2); }; 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) { # authenticate registers if (method=="REGISTER") { if (!radius_www_authorize("mydomain.com")) { www_challenge("mydomain.com", "0"); exit; }; # check the src ip address if(!avp_check("i:2", "eq/$src_ip/ig")) { sl_send_reply("403", "Forbidden IP"); exit; }; save("location"); exit; }; # calls to pstn if(uri=~"sip:00[1-9][0-9]+@") { if(is_method("INVITE") && !has_totag()) { if (!radius_is_user_in("From", "pstn")) { sl_send_reply("403", "Forbidden PSTN"); exit; }; }; # set gateway address rewritehostport("192.168.10.18:5090"); route(1); }; # load callee's avps if(avp_load_radius("callee")) { # check if user has time filter enabled if(avp_check("i:3", "eq/i:1")) { # print time in an avp avp_printf("i:100", "$Tf"); # extract day avp_subst("i:100/i:101", "/(.{3}) .+/*\1*/"); if(!avp_check("i:6", "fm/$day")) { sl_send_reply("403", "Forbidden - day"); exit; }; # extract 'hours:minutes' avp_subst("i:100/i:102", "/(.{10}) (.{5}):.+/\2/"); if((is_avp_set("i:4") && avp_check("i:4", "gt/$time")) || (is_avp_set("i:5") && avp_check("i:5", "lt/$time"))) { sl_send_reply("403", "Forbidden - time"); exit; }; }; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { # log to acc as missed call acc_rad_request("404 Not Found"); acc_log_request("404 Not Found"); sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } # generic forward route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } # ============================================================================ ==================== Users file on RADIUS: ### --- avps --- 101(a)mydomain.com Auth-Type := Accept, Service-Type == "SIP-Callee-AVPs" Sip-Avp += "#3#1", Sip-Avp += "#4:08:00", Sip-Avp += "#5:16:00", Sip-Avp += "#6:Mon,Wed,Thu,Fri" 102(a)mydomain.com Auth-Type := Accept, Service-Type == "SIP-Callee-AVPs" Sip-Avp += "#3#1", Sip-Avp += "#4:08:00", Sip-Avp += "#5:16:00", Sip-Avp += "#6:Mon,Wed,Thu,Free" DEFAULT Auth-Type := Accept, Service-Type == "SIP-Callee-AVPs" ### --- group checking --- ### --- user 101 --- 101(a)mydomain.com Auth-Type := Accept, Sip-Group == "voip", Service-Type == "Group-Check" Reply-Message = "Authorized" 101(a)mydomain.com Auth-Type := Accept, Sip-Group == "pstn", Service-Type == "Group-Check" Reply-Message = "Authorized" ### --- user 102 --- 102(a)mydomain.com Auth-Type := Accept, Sip-Group == "voip", Service-Type == "Group-Check" Reply-Message = "Authorized" DEFAULT Auth-Type := Reject, Service-Type == "Group-Check" ### --- user authentication --- 101(a)mydomain.com Auth-Type := Digest, User-Password == "101" Reply-Message = "Authenticated", Sip-Avp += "rpid:101", Sip-Avp += "#2:192.168.10.17", # Sip-Avp += "#2:192.168.2.11" 102(a)mydomain.com Auth-Type := Digest, User-Password == "102" Reply-Message = "Authenticated", Sip-Avp += "rpid:102", # Sip-Avp += "#2:192.168.2.12" ============================================================================ ==================== And bellow is the dubug from radiusd -X and openser -E -ddddddd: OpenSER Debug: ***************************** 0(4084) SIP Request: 0(4084) method: <REGISTER> 0(4084) uri: <sip:mydomain.com> 0(4084) version: <SIP/2.0> 0(4084) parse_headers: flags=2 0(4084) DEBUG:parse_to:end of header reached, state=8 0(4084) DEBUG: get_hdr_field: <To> [19]; uri=[sip:101@mydomain.com] 0(4084) DEBUG: to body [sip:101@mydomain.com ] 0(4084) get_hdr_field: cseq <Cseq>: <2> <REGISTER> 0(4084) Found param type 232, <branch> = <z9hG4bK-eb137d5b6c8524d7c778e5629555a97e>; state=16 0(4084) end of header reached, state=5 0(4084) parse_headers: Via found, flags=2 0(4084) parse_headers: this is the first via 0(4084) After parse_msg... 0(4084) preparing to run routing scripts... 0(4084) DEBUG:maxfwd:is_maxfwd_present: value = 20 0(4084) DEBUG: add_param: tag=30167a95 0(4084) DEBUG:parse_to:end of header reached, state=29 0(4084) radius_is_user_in(): Failure 0(4084) parse_headers: flags=200 0(4084) DEBUG: get_hdr_body : content_length=0 0(4084) found end of header 0(4084) find_first_route: No Route headers found 0(4084) loose_route: There is no Route HF 0(4084) grep_sock_info - checking if host==us: 9==10 && [mydomain.com] == [192.168.10.18] 0(4084) grep_sock_info - checking if port 5060 matches port 5060 0(4084) grep_sock_info - checking if host==us: 9==10 && [mydomain.com] == [192.168.10.18] 0(4084) grep_sock_info - checking if port 5060 matches port 5060 0(4084) check_nonce(): comparing [448fe8934593e3e7780fffee019ce58577231baf] and [448fe8934593e3e7780fffee019ce58577231baf] 0(4084) DEBUG:auth_radius:radius_authorize_sterman: Success 0(4084) DEBUG:auth_radius:generate_avps: getting SIP AVPs from avpair 225 0(4084) DEBUG:auth_radius:extract_avp: string is <rpid:101> 0(4084) DEBUG:auth_radius:extract_avp: AVP name is <rpid> 0(4084) DEBUG:auth_radius:extract_avp: AVP val is <101> 0(4084) DEBUG:auth_radius:generate_avps: AVP 'rpid'/0='101'/0 has been added 0(4084) DEBUG:auth_radius:extract_avp: string is <#2:192.168.10.17> 0(4084) DEBUG:auth_radius:extract_avp: AVP name is <2> 0(4084) DEBUG:auth_radius:extract_avp: AVP val is <192.168.10.17> 0(4084) DEBUG:auth_radius:generate_avps: AVP 'null'/2='192.168.10.17'/0 has been added 0(4084) DEBUG:avpops:check_avp: check <192.168.10.17> against <192.168.10.17> as str 0(4084) parse_headers: flags=ffffffffffffffff 0(4084) parse_headers: flags=4000000 0(4084) build_contact(): Created Contact HF: Contact: <sip:101@192.168.10.17;LINEID=6f5beb97d08670a1c8328ce1526ae928>;expires=300 0(4084) parse_headers: flags=ffffffffffffffff 0(4084) check_via_address(192.168.10.17, 192.168.10.17, 0) 0(4084) DEBUG:destroy_avp_list: destroying list 0xb60f0a50 0(4084) receive_msg: cleaning up 1(4086) build_recv_sock: <192.168.10.18_5060> ******************************* RADIUS Debug: ******************************* rad_recv: Access-Request packet from host 127.0.0.1:32900, id=196, length=185 User-Name = "101(a)shatel.ir" Digest-Attributes = 0x0a05313031 Digest-Attributes = 0x010b73686174656c2e6972 Digest-Attributes = 0x022a3434386666316666393332663830393337613034373266333837363163353036656166 3636613934 Digest-Attributes = 0x040f7369703a73686174656c2e6972 Digest-Attributes = 0x030a5245474953544552 Digest-Response = "7a3f00f697286dd95c5aa654a9662dea" Service-Type = Sip-Session Sip-Uri-User = "101" NAS-Port = 5060 NAS-IP-Address = 127.0.0.1 Processing the authorize section of radiusd.conf modcall: entering group authorize for request 100 modcall[authorize]: module "preprocess" returns ok for request 100 modcall[authorize]: module "chap" returns noop for request 100 modcall[authorize]: module "mschap" returns noop for request 100 rlm_digest: Adding Auth-Type = DIGEST modcall[authorize]: module "digest" returns ok for request 100 rlm_realm: Looking up realm "shatel.ir" for User-Name = "101(a)shatel.ir" rlm_realm: No such realm "shatel.ir" modcall[authorize]: module "suffix" returns noop for request 100 rlm_eap: No EAP-Message, not doing EAP modcall[authorize]: module "eap" returns noop for request 100 users: Matched entry DEFAULT at line 152 users: Matched entry 101(a)shatel.ir at line 255 modcall[authorize]: module "files" returns ok for request 100 modcall: leaving group authorize (returns ok) for request 100 rad_check_password: Found Auth-Type Digest auth: type "digest" Processing the authenticate section of radiusd.conf modcall: entering group authenticate for request 100 rlm_digest: Converting Digest-Attributes to something sane... Digest-User-Name = "101" Digest-Realm = "shatel.ir" Digest-Nonce = "448ff1ff932f80937a0472f38761c506eaf66a94" Digest-URI = "sip:shatel.ir" Digest-Method = "REGISTER" A1 = 101:shatel.ir:101 A2 = REGISTER:sip:shatel.ir H(A1) = a5d8cf73217ea46c8fd8ca5da1c6a2e3 H(A2) = 27e9df2d1a7ed715c7204d1316039120 KD = a5d8cf73217ea46c8fd8ca5da1c6a2e3:448ff1ff932f80937a0472f38761c506eaf66a94:27 e9df2d1a7ed715c7204d1316039120 EXPECTED 7a3f00f697286dd95c5aa654a9662dea RECEIVED 7a3f00f697286dd95c5aa654a9662dea modcall[authenticate]: module "digest" returns ok for request 100 modcall: leaving group authenticate (returns ok) for request 100 radius_xlat: 'Authenticated' Sending Access-Accept of id 196 to 127.0.0.1 port 32900 Reply-Message = "Authenticated" SIP-AVP += "rpid:101" SIP-AVP += "#2:85.15.7.17" Finished request 100 Going to the next request Waking up in 2 seconds... rad_recv: Access-Request packet from host 127.0.0.1:32901, id=197, length=64 User-Name = "101(a)shatel.ir" Sip-Group = "suspended" Service-Type = Group-Check NAS-Port = 0 NAS-IP-Address = 127.0.0.1 Processing the authorize section of radiusd.conf modcall: entering group authorize for request 101 modcall[authorize]: module "preprocess" returns ok for request 101 modcall[authorize]: module "chap" returns noop for request 101 modcall[authorize]: module "mschap" returns noop for request 101 modcall[authorize]: module "digest" returns noop for request 101 rlm_realm: Looking up realm "shatel.ir" for User-Name = "101(a)shatel.ir" rlm_realm: No such realm "shatel.ir" modcall[authorize]: module "suffix" returns noop for request 101 rlm_eap: No EAP-Message, not doing EAP modcall[authorize]: module "eap" returns noop for request 101 users: Matched entry DEFAULT at line 152 users: Matched entry DEFAULT at line 252 modcall[authorize]: module "files" returns ok for request 101 modcall: leaving group authorize (returns ok) for request 101 rad_check_password: Found Auth-Type Reject rad_check_password: Auth-Type = Reject, rejecting user auth: Failed to validate the user. Delaying request 101 for 1 seconds Finished request 101 Any thoughts? Thanks, Hamid

3 5

[Serusers] Error message: Header V3 DSA signature:
by S.A.S. Lorensuhewa 19 Jun '06

19 Jun '06

Dear All, I got following error message while installing ser rpm. What is the meaning of this. arning: ser-0.9.6-6.fc5.i386.rpm: Header V3 DSA signature: NOKEY, key ID 1ac70ce6 Aruna ---------------------------------------------------------- S. A. S. Lorensuhewa Head, Department of Computer Science University of Ruhuna, Matara, Sri Lanka Telephone: 0094 (0)41 2227019 Fax: 0094 (0)41 2222683 Email: lorensuhewa(a)cc.ruh.ac.lk, aruna(a)cc.ruh.ac.lk

2 1

[Serusers] Unsupported document format
by Alois Schafferhofer 19 Jun '06

19 Jun '06

Hi, I use Presence Snapshot 4.1, and I want my Server to store x and y coordinate and send NOTIFY to the SUBSCRIBER whenever the position changes. I tried it with geopriv location but I get the answer: "Unsupported document format". Is there an error in my PUBLISH request or isn't it possible to store x and y? And in which table does it store? I added the collumns x, y in presentity, but SER ignores them. I hope, you can help me. Thanks. /usr/sbin/ser[9955]: handle publish request /usr/sbin/ser[9955]: ERROR: pidf.c:467: can't parse document /usr/sbin/ser[9955]: can't parse CPIM-PIDF document /usr/sbin/ser[9955]: handled publish U 10.0.0.3:5060 -> 10.0.0.11:5060 PUBLISH sip:PDA_User@fh-joanneum.at SIP/2.0..Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK1845246585..Max-Forwards: 70..From: <sip:PDA_User@fhjoanneu m.at>;tag=1986441488..To: <sip:PDA_User@fh-joanneum.at>..Call-ID: 1734692618- 1859786556-1651516169..CSeq: 6 PUBLISH..Contact: <sip:PDA_User@10.0.0.3>. .Content-Type: application/cpim-pidf+xml..Content-Length: 307..Expires: 7200..Event: presence..User-Agent: SIP .NET 1.0 evaluation version, www.independentsoft.com.... <?xml version="1.0 encoding="utf-8"?> <presence xmlns="urn:ietf:params:xml:ns:pidf"entity="sip:PDA_User@fh-joanneum.at"> <tuple id=9r28r59> <status> <basic>online</basic> <geopriv><locationinfo><civilAddress><x>01525.0125</x><y>04704.2002</y></civilAddress></locationinfo></geopriv> </status> </tuple> </presence> U 10.0.0.11:5060 -> 10.0.0.3:5060 SIP/2.0 415 Unsupported document format..Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK1845246585..From: <sip:PDA_User@fhjoanneum.at>;tag=1986441488.. To: <sip:PDA_User@fh-joanneum.at>;tag=589230a54cd08b3768529724501fae28-a814..Call-ID: 1734692618-1859786556-1651516169..CSeq: 6 PUBLISH..Server: Sip E Xpress router (0.10.99-dev35-pa-4.1 (i386/linux))..Content-Length: 0..Warning: 392 10.0.0.11:5060 "Noisy feedback tells: pid=9955 req_src_ip=10.0.0.3 req_src_port=5060 in_uri=sip:PDA_User@fh-joanneum.at out_uri=sip:PDA_User@fhjoanneum. at via_cnt==1" Regards, Alois

2 1

[Users] invalid port returned by mediaproxy
by Kenny Chua 19 Jun '06

19 Jun '06

Hello, I'm trying to implement mediaproxy with my openser. I'm getting this error: local /usr/local/sbin/openser[2781]: error: use_media_proxy(): invalid port returned by mediaproxy: # I've forwarded all ports that are required. I'm having problems with sip2sip (no audio). Also, for SIP to PSTN, I'm having problems too. The call goes through the PSTN gateway and actually connects. However, its still one way audio or sometimes no audio at all. I'm using FC5 with openser 1.0.1. Mediaproxy 1.7.2. Below are my configuration files for media proxy and openser.cfg: **************************************Openser.cfg********************************* debug=3 fork=yes log_stderror=no listen=192.168.1.5 # This is where OpenSER installed port=5060 children=4 alias=mydomain.com # This is where OpenSER installed dns=no rev_dns=no fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/acc.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", 1) modparam("domain", "db_mode", 1) modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 180) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/openser/sip-asymmetric-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/openser/rtp-asymmetric-clients") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) modparam("registrar", "default_expires", 60) modparam("registrar", "min_expires", 30) modparam("registrar", "nat_flag", 6) modparam("registrar", "use_domain", 1) modparam("rr", "enable_full_lr", 1) modparam("acc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("acc", "failed_transaction_flag", 1) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "db_flag", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); exit; }; # ----------------------------------------------------------------- # Record Route Section and Acc section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { record_route_preset("192.168.1.5:5060;nat=yes"); # This is where OpenSER installed } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); return; }; if (uri==myself) { if (method=="CANCEL") { route(3); return; } else if (method=="INVITE") { xlog("URI is myself, forward to route 3\n"); route(3); return; } else if (method=="REGISTER") { route(2); return; }; lookup("aliases"); if (uri!=myself) { route(1); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("mydomain.com","subscriber")) { www_challenge("mydomain.com","0"); exit; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); exit; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- xlog("Call Invite\n"); if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; lookup("aliases"); if (uri=~"^sip:1[0-9]*@*") { xlog("PSTN Gateway\n"); use_media_proxy(); rewritehostport("pstn.gateway.com:5060"); # This is where PSTN route(1); exit; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; if (method=="CANCEL") { route(1); exit; }; if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; route(1); } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:\ +0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } ***************************Mediaproxy.ini****************************************** [Dispatcher] ;start = yes socket = /var/run/proxydispatcher.sock ;group = ser defaultProxy = /var/run/mediaproxy.sock [MediaProxy] start = yes socket = /var/run/mediaproxy.sock group = ser listen = None allow = None proxyIP = 192.168.1.5 # This is where OpenSER installed portRange = 21000:35000 # These values are set same in ATA's configs for RTP Range TOS = 0xb8 idleTimeout = 60 holdTimeout = 180 forceClose = 0 accounting = off [Accounting] user = root password = host = localhost database = ser table = acc Can anyone figure out my problem? Thank you in advance __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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