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[Serusers] More call transfer questions
by Steve Blair 28 Feb '05

28 Feb '05

I'm a bit stuck and wonder if someone on this list can help. When a call arrives at a Cisco IP phone registered to our SER proxy from the PSTN it is answered. The called party then tries to transfer the call. In either case a new invite is generated. I'm under the impression a refer message should be generated but given the way attended transfer works on the 7960s it doesn't seem possible to have a refer generated. Has anyone else seen this? If so any thoughts on what is happening? Thanks,Steve

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[Serusers] "3xx Redirection" - How Should This Work?
by Java Rockx 28 Feb '05

28 Feb '05

Hi all. I'm trying to get my call working working and I have a question. Suppose a SIP subscriber has voicemail and call forwarding is enabled. When an incoming call is destined for this SIP user, SER will reply with a 302 redirect message. Next the caller's phone should construct a brand new INVITE to the redirected destination. My question is this: If the forwarded destination is busy, then the call should be sent to the voicemail of the SIP subscriber that was originally called. This is the way the RBOCs call forwarding works. If I'm using sl_send_reply("302", "Temporarily Moved") back to the caller, how can I make the voicemail piece work? Regards, Paul

2 3

[Serusers] lookup not finding local users
by Glenn Dalgliesh 28 Feb '05

28 Feb '05

I dont seem to be able to get a match when I send a call from my gw to ser registered user. Debug indicates that user Not found in usrloc even though the database shows the users registered. I am able to make out bound calls the gw just fine. If anyone can shed some light in it it would be great. Part 1 of 2 Part to follow with ser.cfg Thanks ============================================================================ ================================= mysql> select * from location; +----------+--------+------------------------+----------+------------------- --+-------+----------------------------------+------+----------------+------ -----+-------+-------+----------------------------+ | username | domain | contact | received | expires | q | callid | cseq | last_modified | replicate | state | flags | user_agent | +----------+--------+------------------------+----------+------------------- --+-------+----------------------------------+------+----------------+------ -----+-------+-------+----------------------------+ | 9901 | | sip:9901@69.250.205.31 | NULL | 2005-02-26 03:31:09 | -1.00 | e5201b3819008718(a)192.168.199.102 | 107 | 20050226033016 | 0 | 0 | 0 | Grandstream BT100 1.0.5.11 | +----------+--------+------------------------+----------+------------------- --+-------+----------------------------------+------+----------------+------ -----+-------+-------+----------------------------+ 1 row in set (0.01 sec) ====debugt output====================================================================== ========================= 20(18672) parse_headers: flags=256 20(18672) find_first_route: No Route headers found 20(18672) loose_route: There is no Route HF 20(18672) lookup(): '9901' Not found in usrloc ============================================================================ ================================= U 69.19.88.16:5060 -> 69.19.88.19:5060 INVITE sip:9901@69.19.88.19 SIP/2.0..Via: SIP/2.0/UDP 69.19.88.16:5060;branch=z9hG4bK2fb13d17..From: "Cell Phone MD" <sip:4103532264 @69.19.88.16>;tag=as149fbe93..To: <sip:9901@69.19.88.19>..Contact: <sip:4103532264@69.19.88.16>..Call-ID: 745bcf587074e77d4a8118dc212 ccba7@69.19.88.16..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Date: Sat, 26 Feb 2005 08:41:01 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS , BYE, REFER..Content-Type: application/sdp..Content-Length: 240....v=0..o=root 11581 11581 IN IP4 69.19.88.16..s=session..c=IN IP4 66. 159.88.16..t=0 0..m=audio 14934 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:1 01 0-16..a=silenceSupp:off - - - -.. # U 69.19.88.19:5060 -> 69.19.88.16:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 69.19.88.16:5060;branch=z9hG4bK2fb13d17..From: "Cell Phone MD" < sip:4103532264@69.19.88.16>;tag=as149fbe93..To: <sip:9901@69.19.88.19>..Call-ID: 745bcf587074e77d4a8118dc212ccba7@69.19.88.16..CSeq: 102 INVITE..Server: Sip EXpress router (0.9.0 (i386/linux))..Content-Length: 0..Warning: 392 69.19.88.19:5060 "Noisy feedback tells: p id=18682 req_src_ip=69.19.88.16 req_src_port=5060 in_uri=sip:9901@69.19.88.19 out_uri=sip:9901@69.19.88.16:5060 via_cnt==1".... # U 69.19.88.19:5060 -> 69.19.88.16:5060 INVITE sip:9901@69.19.88.16:5060 SIP/2.0..Max-Forwards: 10..Record-Route: <sip:69.19.88.19;ftag=as149fbe93;lr=on>..Via: SIP/2.0/UDP 66 .159.88.19;branch=z9hG4bK96c9.80b8ddd7.0..Via: SIP/2.0/UDP 69.19.88.16:5060;branch=z9hG4bK2fb13d17..From: "Cell Phone MD" <sip:410353 2264(a)69.19.88.16>;tag=as149fbe93..To: <sip:9901@69.19.88.19>..Contact: <sip:4103532264@69.19.88.16>..Call-ID: 745bcf587074e77d4a8118d c212ccba7@69.19.88.16..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Date: Sat, 26 Feb 2005 08:41:01 GMT..Allow: INVITE, ACK, CANCEL, OPT IONS, BYE, REFER..Content-Type: application/sdp..Content-Length: 240..P-hint: GATEWAY....v=0..o=root 11581 11581 IN IP4 69.19.88.16..s= session..c=IN IP4 69.19.88.16..t=0 0..m=audio 14934 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telephone -event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -.. # U 69.19.88.16:5060 -> 69.19.88.19:5060 SIP/2.0 482 Loop Detected..Via: SIP/2.0/UDP 69.19.88.19;branch=z9hG4bK96c9.80b8ddd7.0..Via: SIP/2.0/UDP 69.19.88.16:5060;branch=z9hG4b K2fb13d17..From: "Cell Phone MD" <sip:4103532264@69.19.88.16>;tag=as149fbe93..To: <sip:9901@69.19.88.19>;tag=as149fbe93..Call-ID: 74 5bcf587074e77d4a8118dc212ccba7@69.19.88.16..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER ..Contact: <sip:4103532264@69.19.88.16>..Content-Length: 0....

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[Serusers] SER MySQL + NAT problems
by Paulo Vicentini 28 Feb '05

28 Feb '05

Hi Would mind sending your ser.cfg file, i am facing problems with NAT traversal I have already read > <http://www.informatik.uni-bremen.de/~prelle/terena/cookbook/Co> okbook_D2/ but it still does not working This is my ser.cfg file: Is there anything wrong whit it? # # $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=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ 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 ---------------------------------- loadmodule "/usr/local/lib/ser/modules/nathelper.so" # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # We will you flag 6 to mark NATed contacts modparam("registrar", "nat_flag", 6) # Enable NAT pinging modparam("nathelper", "natping_interval", 60) # Ping only contacts that are known to be # behind NAT modparam("nathelper", "ping_nated_only", 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"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # 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 used); also, # the received test should, if complete, should check all # vias for presence 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 smart enough to be symmetric. In some phones, like # it takes a configuration option. With Cisco 7960, it is # called NAT_Enable=Yes, with kphone it is called # "symmetric media" and "symmetric signaling". (The latter # not part of public released yet.) 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 record_route(); # loose-route processing if (loose_route()) { t_relay(); 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("200.151.192.13", "subscriber")) { www_challenge("200.151.192.13", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } # # Forcing media relay if necessary # route[1] { if (uri=~"[@:](192\.168\.|10\.|172\.16)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; if (isflagset(6)) { force_rtp_proxy(); t_on_reply("1"); append_hf("P-Behind-NAT: Yes\r\n"); }; if (!t_relay()) { sl_reply_error(); break; }; } onreply_route[1] { if (status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); }; } Thanks Paulo _________________________________________________________________ MSN Messenger: converse online com seus amigos . http://messenger.msn.com.br

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[Serusers] ERROR: tcp_blocking_connect !!!
by Mohammad Khan 28 Feb '05

28 Feb '05

Hello List: I am getting this error message: ERROR: tcp_blocking_connect: timeout (10) By doing a search in Archive, I have found, it tcp connection take long time, this error raise. Is it right? I am not aware about any network problem in my network. Other issue, If any UA register, my locatoin table keeps record very good. But if UA re-register, location table have double entry of that user. Any idea, why its happening or how to fix it. BTW, I am running ser-0.9.0 with cvs version of rtpproxy. Thanks, MOhammad

1 0

[Serusers] SIP encrypted calls
by Stefano 28 Feb '05

28 Feb '05

Hi all, i have a question: -do you know how it's possible to encrypt the comunication between two SIP client? -does the proxy implement some particular tasks or actions, or is it a client-side problem? -if it is so, do you know any SIP client that implements encription? Thank you very much stefano __________________________________________________________________ Tiscali Adsl 3 Mega Flat, 3 MESI GRATIS! Con Tiscali Adsl 3 Mega Flat navighi in Rete alla supervelocita' a soli 29.95 euro al mese senza limiti di tempo. Attivati entro il 28 Febbraio 2005, 3 MESI sono GRATIS Scopri come http://abbonati.tiscali.it/adsl/sa/2flat_tc/

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[Serusers] part 2 lookup not finding local users
by Glenn Dalgliesh 28 Feb '05

28 Feb '05

Part 2 of 2 ==ser.cfg=================================================================== ======================================== # # $Id: serconf.sh,v 1.3 2003/11/03 13:03:58 janakj Exp $ # # autogenerated SER configuration # # user: uid=0(root) gid=0(root) groups=0(root),1(bin),2(daemon),3(sys),4(adm),6(disk),10(wheel) # system: Linux baltodb.popdial.com 2.6.10-1.760_FC3 #1 Wed Feb 2 00:14:23 EST 2005 i686 i686 i386 GNU/Linux # date: Sat Feb 26 02:57:50 EST 2005 # # ----------- global configuration parameters ------------------------ /* debug=4 fork=yes port=5060 log_stderror=no memlog=5 */ mhomed=yes fifo="/tmp/ser_fifo" alias=baltodb.popdial.com # uncomment to override config values for test # /* debug=4 # debug level (cmd line: -ddd) fork=yes port=5060 log_stderror=yes # (cmd line: -E) fifo="/tmp/ser_fifox" # */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) children=16 # if changing fifo mode to a more restrictive value, put # decimal value in there, e.g. dec(rw|rw|rw)=dec(666)=438 #fifo_mode=438 # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" loadmodule "/usr/local/lib/ser/modules/enum.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" # ----------------- setting module-specific parameters --------------- # all DB urls here #modparam("usrloc|acc|auth_db|group|msilo|uri", "db_url","mysql://ser:heslo@localhost/ser") modparam("usrloc|acc|auth_db|group|msilo|uri_db", "db_url","mysql://ser:heslo@localhost/ser") # -- usrloc params -- /* 0 -- dont use mysql, 1 -- write_through, 2--write_back */ modparam("usrloc", "db_mode", 2) modparam("usrloc", "timer_interval", 10) # -- auth params -- modparam("auth_db", "calculate_ha1", yes) #modparam("auth_db", "user_column", "user_id") modparam("auth_db", "password_column", "password") modparam("auth", "nonce_expire", 300) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- acc params -- # that is the flag for which we will account -- don't forget to modparam("acc", "db_flag", 1 ) modparam("acc", "db_missed_flag", 3 ) # -- tm params -- modparam("tm", "fr_timer", 20 ) modparam("tm", "fr_inv_timer", 90 ) modparam("tm", "wt_timer", 20 ) # -- msilo params modparam("msilo", "registrar", "sip:registrar@baltodb.popdial.com") # -- enum params -- # modparam("enum", "domain_suffix", "e164.arpa.") # ------------------------- request routing logic ------------------- # main routing logic route{ /* ********* ROUTINE CHECKS ********************************** */ # filter too old messages if (!mf_process_maxfwd_header("10")) { log("LOG: Too many hops\n"); sl_send_reply("483","Alas Too Many Hops"); break; }; #if (len_gt( max_len )) { if ( msg:len > max_len ) { sl_send_reply("513", "Message too large sorry"); break; }; # Make sure that requests dont advertise addresses # from private IP space (RFC1918) in Contact HF # (note: does not match with folded lines) if (search("^(Contact|m): .*(a)(192\.168\.|10\.|172\.16)")) { # 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:")) && !( src_ip==192.168.0.0/16 || src_ip==10.0.0.0/8 || src_ip==172.16.0.0/12 )) { log("LOG: Someone trying to register from private IP again\n"); sl_send_reply("479", "We dont accept private IP contacts" ); break; }; }; # anti-spam -- if somene claims to belong to our domain in From, # challenge him (skip REGISTERs -- we will chalenge them later) if (search("(From|F):.*(baltodb\.popdial\.com|66\.159\.88\.19)")) { # invites forwarded to other domains, like FWD may cause subsequent # request to come from there but have iptel in From -> verify # only INVITEs (ignore FIFO/UAC's requests, i.e. src_ip==myself) if (method=="INVITE" & !(src_ip==69.19.88.19)) { if (!(proxy_authorize( "baltodb.popdial.com" /* realm */, "subscriber" /* table name */ ))) { proxy_challenge("baltodb.popdial.com" /* realm */, "0" /* no-qop */); break; }; # to maintain outside credibility of our proxy, we enforce # username in From to equal digest username; user with # "john.doe" id could advertise "bill.gates" in From otherwise; if (!check_from()) { log("LOG: From Cheating attempt in INVITE\n"); sl_send_reply("403", "That is ugly -- use From=id next time (OB)"); break; }; # we better don't consume credentials -- some requests may be # spiraled through our server (sfo@iptel->7141@iptel) and the # subsequent iteration may challenge too, for example because of # iptel claim in From; UACs then give up because they # already submitted credentials for the given realm #consume_credentials(); }; # INVITEs claiming to come from our domain } else if (method=="INVITE" && !(uri=~"[@:\.](baltodb\.popdial\.com|66\.159\.88\.19)([;:].*)*" # ... and we serve our gateway too if present | uri=~"@66\.159\.88\.16([;:].*)*" )) { #the INVITE neither claims to come from our domain nor is it targeted to it # -> junk it sl_send_reply("403", "No relaying"); break; }; /* ********* RR ********************************** */ # to be safe, record route everything; UAs may use different # transport protocols and need to have SER in path record_route(); # if route forces us to forward to some explicit destination, # do so; check however first that a cheater didn't preload # a gateway destination to bypass PSTN ACLs if (loose_route()) { if (uri=~"sip:[+0-9]+@66\.159\.88\.16") { # it is gateway -- proceed to ACLs route(3); break; }; # route HF determined next hop; forward there append_hf("P-hint: rr-enforced\r\n"); t_relay(); break; }; /* ********* check for requests targeted out of our domain... ******* */ # sign of our domain: there is '@' (username) or : (nothing) in # front of our domain name ; ('.' is not there -- we handle all # xxx.iptel.org as outbound hosts);if none of these cases matches, # proceed with processing of outbound requests in route[2] if (!(uri=~"[@:](sip[\.)?(baltodb\.popdial\.com|66\.159\.88\.19)([;:].*)*" | uri=~"@66\.159\.88\.16([;:].*)*")) { route(2); break; }; /* ************ requests for our domain ********** */ /* now, the request is for sure for our domain */ # registers always MUST be authenticated to # avoid stealing incoming calls if (method=="REGISTER") { # Make sure that user's dont register infinite loops # (note: does not match with folded lines) if (search("^(Contact|m): .*(a)(baltodb\.popdial\.com|66\.159\.88\.19)")) { log(1, "LOG: alert: someone trying to set aor==contact\n"); sl_send_reply("476", "No Server Address in Contacts Allowed" ); break; }; if (search("^(Contact|m): .*66\.159\.88\.16")) { log(1, "LOG: alert: protected contacts\n"); sl_send_reply("476", "No Server Address in Contacts Allowed" ); break; }; if (!www_authorize( "baltodb.popdial.com" /* realm */, "subscriber" /* table name */ )) { # challenge if none or invalid credentials www_challenge( "baltodb.popdial.com" /* realm */, "0" /* no qop -- some phones can't deal with it */); break; }; # prohibit attempts to grab someone else's To address # using valid credentials; if (!check_to()) { log("LOG: To Cheating attempt\n"); sl_send_reply("403", "That is ugly -- use To=id in REGISTERs"); break; }; # it is an authenticated request, update Contact database now if (!save("location")) { sl_reply_error(); }; m_dump(); break; }; # some UACs might be fooled by Contacts our UACs generate to make MSN # happy (web-im, e.g.) -- tell its urneachable if (uri=~"sip:daemon@" ) { sl_send_reply("410", "daemon is gone"); break; }; # is this an ENUM destination (leading +?)? give it a try, if the lookup # doesn't change URI, just continue if (uri=~"sip:\+[0-9]+@") { if (!enum_query("voice")) { # if parameter empty, it defaults to "e2u+sip" enum_query(""); # E2U+sip }; } else { # aliases (take precedences over PSTN number; provisioning interface # is set up to assinge aliases beginning with 8) lookup("aliases"); }; # check again, if it is still for our domain after aliases are resolved if (!(uri=~"[@:](sip[\.)?(baltodb\.popdial\.com|66\.159\.88\.19)([;:].*)*" | uri=~"@66\.159\.88\.16([;:].*)*")) { route(5); break; }; # now check if it's about PSTN destinations through our gateway; # note that 8.... is exempted for numerical non-gw destinations if (uri=~"sip:\+?[0-79][0-9]*@.*") { route(3); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { # handle user which was not found ... route(4); break; }; # check whether some inventive user has uploaded gateway # contacts to UsrLoc to bypass our authorization logic if (uri=~"sip:[+0-9]+@66\.159\.88\.16") { # it is gateway -- proceed to ACLs route(3); break; }; /* ... and also report on missed calls ... */ setflag(3); # we now know we may, we know where, let it go out now! append_hf("P-hint: USRLOC\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } #------------------- OUTBOUND ---------------------------------------- # routing logic for outbound requests targeted out of our domain # (keep in mind messages to our users can end up here too: for example, # an INVITE may be UsrLoc-ed, then the other party uses outbound # proxy with r-uri=the usr_loced addredd (typically IP)) route[2] { append_hf("P-hint: OUTBOUND\r\n"); t_relay(); } #------- ALIASED OUTBOUND -------------------------------------------- # routing logic for inbound requests aliased outbound; unlike # with real outbound requests we do not force authentication # as these calls are server by our server and we do not want # to disqualify unathenticated request originatiors from other # domains route[5] { append_hf("P-hint: ALIASED-OUTBOUND\r\n"); t_relay(); } #----------------- PSTN ---------------------------------------------- # logic for calls to the PSTN route[3] { # turn accounting on setflag(1); /* require all who call PSTN to be members of the "int" group; apply ACLs only to INVITEs -- we don't need to protect other requests, as they don't imply charges; also it could cause troubles when a call comes in via PSTN and goes to a party that can't authenticate (voicemail, other domain) -- BYEs would fail then; exempt Cisco gateway from authentication by IP address -- it does not support digest */ if (method=="INVITE" && (!src_ip==69.19.88.16)) { if (!proxy_authorize( "baltodb.popdial.com" /* realm */, "subscriber" /* table name */)) { proxy_challenge( "baltodb.popdial.com" /* realm */, "0" /* no qop */ ); break; }; # let's check from=id ... avoids accounting confusion if (method=="INVITE" & !check_from()) { log("LOG: From Cheating attempt\n"); sl_send_reply("403", "That is ugly -- use From=id next time (gw)"); break; }; if(!is_user_in("credentials", "int")) { sl_send_reply("403", "NO PSTN Privileges..."); break; }; consume_credentials(); }; # INVITE to authorized PSTN # if you have passed through all the checks, let your call go to GW! rewritehostport("69.19.88.16:5060"); # snom conditioner if (method=="INVITE" && search("User-Agent: snom")) { replace("100rel, ", ""); }; append_hf("P-hint: GATEWAY\r\n"); # use UDP to guarantee well-known sender port (TCP ephemeral) t_relay_to_udp("69.19.88.16","5060"); } /* *********** handling of unavailable user ******************* */ route[4] { /**/ # message store if (method=="MESSAGE") { t_newtran(); if (m_store("0")) { t_reply("202", "Accepted for Later Delivery"); } else { t_reply("503", "Service Unavailable"); }; break; }; /**/ # non-Voip -- just send "off-line" if (!(method=="INVITE" || method=="ACK" || method=="CANCEL")) { sl_send_reply("404", "Not Found"); break; }; # voicemail subscribers ... t_newtran(); t_reply("404", "Not Found"); # we account missed incoming calls; previous statteful processing # guarantees that retransmissions are not accounted if (method=="INVITE") { acc_db_request("404 missed call", "missed_calls"); }; }

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[Serusers] ser + rtpproxy - call disconnected
by support 28 Feb '05

28 Feb '05

Hi, When I try to make a call using rtpproxy and ser-0.8.14 (SIP UA <---> PSTN), most of the time the call will be disconnected within 1 minutes. Sometimes, it will be so unstable that the call be disconnected after 15 seconds. The above scenarios happens Will this problem related to unstable rtpproxy? Because I know that all packets will route through the server. once rtp packet get loss, the call will be disconnected. Is this correct? Perhaps it depends on my ser.cfg. Thomas My ser.cfg: # # $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=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen="" port=5060 children=4 fifo_mode=0666 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- 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/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- Nathelper params -- modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT # ------------------------- request routing logic ------------------- # main routing logic route{ # ----------------------------------------------- # Sanity Check Section # ----------------------------------------------- # 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; }; # ----------------------------------------------- # NOTIFY Keep-Alive Section # ----------------------------------------------- if ((method=="NOTIFY") && search("^Event: keep-alive")) { sl_send_reply("200","OK"); break; }; # Nathelper 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:")) { 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("", "subscriber")) { www_challenge("", "0"); break; }; save("location"); break; }; # if the dialed number lies in the range 35891500-35891799, don't forward it to T1 Trunk GW if ((uri=~"^sip:(852|)358915[0-9][0-9]@") || (uri=~"^sip:(852|)358916[0-9][0-9]@") || (uri=~"^sip:(852|)358917[0-9][0-9]@")) { if (uri=~"^sip:852*") { strip(3); }; }; 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 # Call Routing Section if (!lookup("location")) { if (uri=~"^sip:(852|)[0-9]{8}@") { # Send to PSTN Gateway route(2); break; }; sl_send_reply("404", "User 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(); break; }; } # PSTN Call to T1 Trunk GW route[2] { rewritehostport(""); if (isflagset(6)) { force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { sl_reply_error(); break; }; } # !! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); # Not all 2xx messages have a content body so here we make sure # out Content-Length > 0 to avoid a parse error if (!search("^Content-Length:\0")) { 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(); }; } ---------end of config -----------

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[Serusers] 0.9.0 release
by Daniel-Constantin Mierla 28 Feb '05

28 Feb '05

The release of 0.9.0 should be finalized very soon. No bugs found for a while -- it might be due to no usage or no other bug there :-) . Would be appreciated any feedback regarding the usage of this version in order to estimate the level of stability. A short reply including how are you using the 0.9.0 (production/testing) and the list with used modules (copy&paste of loadmodule section to save time :-) for those who don't have it) would be very appreciated. Thanks, Daniel PS. If you don't want to make it public, then send it just to me.

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[Serusers] no reply from radius server
by Seong bear 28 Feb '05

28 Feb '05

Hi, I'm trying to startup my ser with radius authentication, however i encountered the same problem with this post: http://lists.iptel.org/pipermail/serusers/2005-January/014425.html. I really appreciate that someone could point me put the problelm. :-( authserver radiusserver, 3.4.5.6 acctserver radiusserver, 3.4.5.6 where radiusserver with ip 3.4.5.6 in /etc/hosts, and tcdump didn't show any traffic with destinated radius server ip or hostname rc_send_server: no reply from RADIUS server radiusserver:1812 --> and sometime, i can't get to see this message. and the server just startup with log below without any message followed by. Feb 28 22:55:44 bear ser: Listening on Feb 28 22:55:44 bear ser: 127.0.0.1 [127.0.0.1]:5080 Feb 28 22:55:44 bear ser: 192.168.0.68 [192.168.0.68]:5080 Feb 28 22:55:44 bear ser: Aliases: localhost:5080 localhost.localdomain:5080 bear:5080 bear.abc.com:5080 Feb 28 22:55:44 bear ser: ser startup succeeded Feb 28 22:55:44 bear ser[15935]: INFO: udp_init: SO_RCVBUF is initially 110592 Feb 28 22:55:44 bear ser[15935]: INFO: udp_init: SO_RCVBUF is finally 221184 Feb 28 22:55:45 bear ser[15935]: INFO: udp_init: SO_RCVBUF is initially 110592 Feb 28 22:55:45 bear ser[15935]: INFO: udp_init: SO_RCVBUF is finally 221184 Feb 28 22:55:45 bear ser[15946]: INFO: fifo process starting: 15946 Feb 28 22:55:45 bear ser[15946]: SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... This is my ser.cfg # ----------- 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 debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5080 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/acc.so" loadmodule "/usr/local/lib/ser/modules/auth.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/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_radius.so" #loadmodule "/usr/local/lib/ser/modules/mysql.so" #loadmodule "/usr/local/lib/ser/modules/im.so" #loadmodule "/usr/local/lib/ser/modules/jabber.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 --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") modparam("auth_radius", "service_type", 15) modparam("auth_radius", "radius_config","/usr/local/etc/radiusclient/radiusclient.conf") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) #modparam("acc", "log_level", 1) modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 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"); 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; }; # 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) { log(1, "Checking uri myself\n"); if (method=="REGISTER") { log(1, "after method REGISTER\n"); # Uncomment this if you want to use digest authentication if (!radius_www_authorize("")) { log(1, "REGISTER: challenging user\n"); www_challenge("", "0"); break; }; save("location"); break; }; if (method=="INVITE") { log(1, "INVITE\n"); setflag(1); /* set for accounting (the same value as in log_flag!) */ }; if (method=="MESSAGE") { log(1, "MESSAGE\n"); setflag(1); /* set for accounting (the same value as in log_flag!) */ }; if (method=="BYE" || method=="CANCEL") { log (1, "BYE or CANCEL\n"); setflag(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; }

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[Serusers] How Can I Change the Contact Header for "302 Temporarily Moved" Responses?
by Java Rockx 28 Feb '05

28 Feb '05

Hi All. I'm working through a call forwarding problem. When call forwarding is enabled on a SIP phone and someone calls that forwarded SIP phone, SER should reply with a 302 reponse code to indicate that the party has been forwarded to a new location. How can I have SER put the new contact location in the response message? I use avpops to read the call forwarding settings, so do I need to use textops to replace the contact header? If so, can I do that in a reply_route or something? Regards, Paul

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Re: [Serusers] SIP encrypted calls
by Cesc Santasusana 28 Feb '05

28 Feb '05

Hi, Full security for SIP calls is well defined, though there are several ways to go. IPSec is always there, though is not very flexible. You can encrypt and authenticate the signalling and the media. Keys can be either manually distributed or dinamycally created using a IKE (defined in the IPSec RFCs). I would not recommend the use of IPSec in a SIP environment, specially for the media. For the media (RTP), the Secure RTP (SRTP) protocol is way better. The overhead added is way smaller than that added to obtain equivalent protection using IPSec (authenticated ESP). Also, it is transparent to media proxies: the SRTP headers are only authenticated, not encrypted; only the body (data) of RTP packet is encrypted; the rest (UDP headers, RTP headers) are left plain. The SRTP keys can be obtained in several ways. The old manual keying method is always there, but there are several other more dynamic. * The k= SDP parameter, which sends a key in plain. This means that SDP needs to be encrypted (S/MIME for end-to-end, or at least TLS on every hop). BTW, i dont like S/MIME :) * The newer k-mgmt= SDP parameter. In this parameter, a full protocol (with embedded authentication and encryption) can be attached as the value, where the keys and SRTP parameters can be securely exchanged. See MIKEY (rfc 3830) and the draft on how to transport it over SDP (draft-ietf-mmusic-kmgmt-ext-xxx, on IETF last call). This provides for end-to-end negotiation of SRTP keys, and i think it is the best way to go. MIKEY is very flexible, suitable for several scenarios. In this scenario, using MIKEY over SDP, where MIKEY is self-protected, it is only left to protect against manipulation of the SIP message: an attacker removing the MIKEY sdp, thus removing security. This can be prevented using TLS on a hop-by-hop basis, if all proxies can be trusted. Again, S/MIME is another option, but i think TLS is better. S/MIME may prevent proxies of inspecting all headers needed during the exchange, whereas TLS would not. As for support of these features ... i know of one softphone supporting the SRTP/MIKEY/TLS approach ... minisip (www.minisip.org). It even has some IPSec support. I've tried, and it works beautifully. The beauty of MIKEY is that it is end-to-end and transparent to proxies, and the negotiation is done in just one round-trip, following the offer-answer SIP model. Very appropriate. Hope it helps, Cesc >>> Nils Ohlmeier <lists(a)ohlmeier.org> 02/28/05 12:44PM >>> Hi Klaus, On Monday 28 February 2005 11:31, Klaus Darilion wrote: > Nils Ohlmeier wrote: > > There are clients. See my previous mail. SRTP is completely transparent > > for SIP proxies. > > How will the clients exchange the key for the RTP encryption? Will it be > sent in the SDP? If yes, wouldn't I also need encrypted SIP to hide the > RTP key? Yes the keys will be exchanged within the SDP. So indeed you should crypt the SDP in signaling either by using TLS or S/MIME. Otherwise someone could read the keys from the signaling and decrypt the RTP streams. Greetings Nils Unclassified _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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Re: [Serusers] SER+Asterisk+NAT. Need a working example or need to talk about it.
by Alexey N. Kovyrin ＠ Home 28 Feb '05

28 Feb '05

Charles Wang wrote: >Dear Alexey: > >Can you send me your config about ser.cfg & sip.conf, extensions.conf >for me to reference? > >My UA1(under NAT) can make a call to Asterisk via SER, then Asterisk >forward the call to PSTN(behind a CISCO 5300). > >But only UA1 can talk to PSTN side, PSTN side can't talk to UA1. > >Here are my sip.conf & extensions.conf. > >I just want to make a call to PSTN using a UA behind NAT. > >Is any config necessary to modify in my goal? > > First of all, you need to add some record for incoming calls from CISCO to sip.conf... [from-cisco] type=peer context=cisco host=61.220.190.243 Next, you need to add [cisco] section to your extensions.conf... [cisco] exten => _., 1, Answer exten => _., 2, Dial(sip/ser/USER_NAME_HERE) exten => _., 3, Hangup Note: It's for all calls be forwarded to same user... If you need (as i think) some more efficient scheme, use more extensions in you extensions.conf... Or do semething like that: [cisco] exten => _XXXXXXXXXX,1,Dial(SIP/${EXTEN}@${SERADDRESS},20,r) -- /Scoundrel

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[Serusers] Unauthencated to Authenticated Call --
by Kannaiyan Natesan 28 Feb '05

28 Feb '05

Hi, I want to forward an unauthenticated call to an authenticated call. Can anyone kindly guide me how can i do the same. UA -- Unauthenticated call -- My Server -- Authenticated Call - Customer I have the authenticated information on my server to handle the customer. Kannaiyan.

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[Serusers] SIP/2.0 404 Not found
by Sun Shung 28 Feb '05

28 Feb '05

Hi everyone, I uses SIPp to test up my SER server and i get the following error. -------------------------------------------------------------------------------------------- 2005-02-28 19:35:50: Unexpected message for Call-ID '36.9402.127.0.0.1(a)sipp.call.id': while expecting '100' response, received 'SIP/2.0 404 Not Found Via: SIP/2.0/UDP 127.0.0.1:5061 From: sipp <sip:sipp@127.0.0.1:5061>;tag=36 To: sut <sip:service@127.0.0.1:5060>;tag=b27e1a1d33761e85846fc98f5f3a7e58.cba5 Call-ID: 36.9402.127.0.0.1(a)sipp.call.id CSeq: 1 INVITE Server: Sip EXpress router (0.8.12 (i386/linux)) Content-Length: 0 Warning: 392 127.0.0.1:5060 "Noisy feedback tells: pid=2833 req_src_ip=127.0.0.1 req_src_port=5061 in_uri=sip:service@127.0.0.1:5060 out_uri=sip:service@127.0.0.1:5060 via_cnt==1" --------------------------------------------------------------------------------------------- When i run SIPp, Its only invite and doesnt hav any other response from the server. I m using the default ser.cfg file. Thanx alot guys From, sunshung _________________________________________________________________ FREE pop-up blocking with the new MSN Toolbar get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/

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[Serusers] uac module
by Elena Ramona Modroiu 28 Feb '05

28 Feb '05

I have just committed a new module "uac" that implements basic SIP UAC functionalities like: - manipulation of From header URI (caller id blocking, anonymization) - client-side authentication (SER should be able to authenticate itself as response to a challenge reply) - hopefully more in the future :-) . This first version has a known limitation in authentication part - qop is not supported. Any feedback regarding the functionality or possible errors is highly appreciated. Ramona

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[Serusers] Redirect responses and accounting
by Thomas Björklund 28 Feb '05

28 Feb '05

Hi everyone, I've been away from SER for a long time but now I have renewed interest. I'm confronted by a problem I've failed to solve by my self, maybe you could help, this is regarding accounting and what happens when a customer responds with a 3xx response (i.e. redirect). Call from customer to proxy works fine. 0(13212) Mon Feb 28 02:38:23 2005 - 130.244.194.233 - INVITE Call-ID: 00036bc3-7aa519a2-610f6e63-7f1e22c2(a)130.244.194.233 From: sip:0856204081@sip-corporate1.testdomain.com To: sip:0890510@sip-corporate1.testdomain.com 0(13212) Routeblock 2 - Calls from customers 0(13212) Call from customer: Test 0(13212) Call passed A-number check 0(13212) Routeblock 3 - To customer? 0(13212) Routeblock 4 - Call not to customer, to gateways 0(13212) Replyroute 1 0(13212) Replyroute 1 0(13212) Replyroute 1 0(13212) ACC: transaction answered: method=INVITE, uid=n/a, call_id=00036bc3-7aa519a2-610f6e63-7f1e22c2(a)130.244.194.233, from="0856204081" <sip:0856204081@sip-corporate1.testdomain.com>;tag=00036bc37aa5007d581f01d7-0ea60293, to=<sip:0890510@sip-corporate1.testdomain.com>;tag=18F3A96C-20F4, i-uri=sip:0890510@sip-corporate1.testdomain.com, o-uri=sip:0890510@sip-gw.swip.net:5060, fromtag=00036bc37aa5007d581f01d7-0ea60293, code=200 0(13212) ACC: transaction answered: method=BYE, uid=n/a, call_id=00036bc3-7aa519a2-610f6e63-7f1e22c2(a)130.244.194.233, from="0856204081" <sip:0856204081@sip-corporate1.testdomain.com>;tag=00036bc37aa5007d581f01d7-0ea60293, to=<sip:0890510@sip-corporate1.testdomain.com>;tag=18F3A96C-20F4, i-uri=sip:0890510@130.244.190.42:5060;ftag=00036bc37aa5007d581f01d7-0ea60293;lr=on, o-uri=sip:0890510@130.244.188.14:5060, fromtag=00036bc37aa5007d581f01d7-0ea60293, code=200 Accounting works fine with START, STOP Call from PSTN to customer, works fine, I have choosen not to generate a START record to make it easier for the billing department. 0(13212) Mon Feb 28 02:41:24 2005 - 130.244.188.14 - INVITE Call-ID: 8CB1DEF-889B11D9-80F1FA83-930B318E(a)130.244.188.14 From: sip:0856264000@130.244.188.14 To: sip:0856204081@130.244.190.42 0(13212) Call from gateway 0(13212) Routeblock 3 - To customer? 0(13212) Call to customer: Test 0(13212) Replyroute 1 0(13212) Replyroute 1 0(13212) Replyroute 1 0(13212) ACC: transaction answered: method=BYE, uid=n/a, call_id=8CB1DEF-889B11D9-80F1FA83-930B318E(a)130.244.188.14, from=<sip:0856264000@130.244.188.14>;tag=18F66B24-226F, to=<sip:0856204081@130.244.190.42>;tag=00036bc37aa5007f1d13f1ad-2475ba78, i-uri=sip:0856204081@130.244.190.42:5060;ftag=18F66B24-226F;lr=on, o-uri=sip:0856204081@130.244.194.233:5060, fromtag=18F66B24-226F, code=200 Here comes the tricky part. Call from PSTN to customer and customer has redirected his extension to a PSTN number. 0(13212) Mon Feb 28 02:42:49 2005 - 130.244.188.14 - INVITE Call-ID: 3B95F3CB-889B11D9-80F5FA83-930B318E(a)130.244.188.14 From: sip:0856264000@130.244.188.14 To: sip:0856204081@130.244.190.42 0(13212) Call from gateway 0(13212) Routeblock 3 - To customer? 0(13212) Call to customer: Test 0(13212) Replyroute 1 0(13212) Redirect prohibited This call generates no accounting (not even with setflag(1) at the top of the route block) I've managed to trigger the 'Redirect prohibited' message by using onreply_route[] but I find no way of canceling the request there. What I want is to be able to either disallow 3xx responses completly or by selectivly cancelling them in onreply_route. Another question, my ser-users archive just broke and I haven't been able to do much searching, what's the best way today to handle several pstn gateways and loadbalace between them for outgoing calls? Best regards, Thomas Björklund

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[Serusers] Mediaproxy: Proxydispatcher hangs
by Martin Koenig 28 Feb '05

28 Feb '05

Hello, last week, we have had issues with a hanging proxydispatcher, which then caused the whole ser setup to hang. Question: Is there a way to "ping" the proxydispatcher process via FIFO, to see whether it is still operating normally? Afaik, the proxydispatcher could be kill -9ed and restartet then without causing major issues for ongoing calls, because the calls are handled by the seperate mediaproxy process? Same question is valid for the mediaproxy process itself, altough "ping" would be TCP/IP or FIFO here. Thank you, with best regards, Martin

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[Serusers] SER+SERWEB+SEMS Deployment
by matt morris 28 Feb '05

28 Feb '05

Hello List, I'm a new user of SER. Just some questions about how I could deploy SER+SERWEB+SEMS, possibly on separate hosts, as I plan to have more than one SER proxy server with a centralized web server and media/voice mail server (1) Basically what features will I be missing if I decide to omit the fifo functions (if possible) in serweb? ( aliases, add new contact SIP address and who are online would be three features, right?) (2) I saw in one of the threads that there is a fifo_server.php file available for fifo-Internet relay, but not quite sure how to use it. Could someone give me some pointers? (3) Somewhere in the mailing list I saw the "fifo_db_url" global declaration, is it a feature only available in version 0.9.0? And is there a serweb version that will work with ser 0.9.0? (4) I have not tried SEMS yet, but since it also communicates with SER via fifo, I would assume it has to be in the same host as SER, right? Are there any existing modules/plugins that will allow SER and SEMS on different hosts? If not, would it be more scalable if I build an Asterisk box to handle voice mails? (5) If I have multiple SER proxy servers on different hosts, how could UA Alpha registered with SER proxy A be able to call UA beta which registered with SER proxy B? What setup/configuration would I need? I know I have some dumb questions - I'm new to voip and SER. Please try to help me out, I would really appreciate it. Thank you all in advance. _________________________________________________________________ Designer Mail isn't just fun to send, it's fun to receive. Use special stationery, fonts and colors. http://join.msn.com/?pgmarket=en-ca&page=byoa/prem&xAPID=1994&DI=1034&SU=ht… Start enjoying all the benefits of MSN® Premium right now and get the first two months FREE*.

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[Serusers] SER and quintum GW for pstn termination
by Noel Paul Evangelista 28 Feb '05

28 Feb '05

good day! i am running ser v.0.8.12 and i am able to make the quintum, the ata and x-lite talk with each other. this is like a UA to UA call. i am using the defualt ser.cfg. how do i configure ser.cfg so that it can make a PSTN call via a quintum gateway that is connected to the TELCO. which part of the ser.cfg will i update. also, on the quintum gateway side, how do i configure it to accept the call and hop-off on the pstn port. i was successful in configuring the quintum to accept calls and ring the pbx port. any help will be greatly appreciated. thanks and regards, noel __________________________________ Do you Yahoo!? Yahoo! Mail - You care about security. So do we. http://promotions.yahoo.com/new_mail

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Re: [Serusers] Manipulate from_uri
by Charles Wang 27 Feb '05

27 Feb '05

Dear Paul: I am a new user of SER sip. And would you please tell me an explain about what the RPID is?? I want to control the call to PSTN with B2BUA of vovida or Asterisk. Best Regard Charles On Wed, 23 Feb 2005 10:33:26 -0500, Java Rockx <javarockx(a)gmail.com> wrote: > Vitaly, > > I believe you are correct that RPID is only helpful during SIP->PSTN calls. > > I'm not real familiar with B2BUA, but if I were to introduce one in to > my system, does that mean I need to have all the RTP traffic pass > through my network as well, or does the B2BUA only deal with SIP > messaging? > > Regards, > Paul > > On Wed, 23 Feb 2005 10:14:18 -0500, Vitaly Nikolaev > <vitaly(a)voipsonic.com> wrote: > > > > The answer is B2BUA > > > > There actually two ways, > > > > First is b2bua which is nice (for me ok.. I like it :) > > > > Second if you assign the CallerID number to device.. the real callerid, (for > > example in my network we use some fake numbers into devices and then change > > it do real callerid (did) in b2bua), then some devices, for example SIPURA, > > can send Anonymous as name in From and Contact and some GWs, will use it and > > hide callerid on outgoing call. > > > > And third: > > > > And there are also Remote-Party-Id: field in SIP that partially supported by > > SER and supported by most of GW and carriers, and that is probably RIGHT way > > to do that. > > > > When PSTN GW sends call to outside of ur network, it no just skip callerid, > > u can do it but it is wrong, it make it private, so usual ppl will not see > > it but for example 911 and some toll free subscriber will see. That right > > way and that how everybody should do cli blocking. > > > > What is wrong in this method is that if you call goes from SIP to SIP and > > you use as me fake CLI on devices, customers will see fake numbers because > > as far as I know SIP devices does not support RPID (please correct me if I > > am wrong) so in my case I will have to use combination of N1 and N3.. > > > > > > -----Original Message----- > > From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On > > Behalf Of Java Rockx > > Sent: Wednesday, February 23, 2005 7:20 AM > > To: Klaus Darilion > > Cc: serusers(a)lists.iptel.org > > Subject: Re: [Serusers] Manipulate from_uri > > > > Klaus, > > > > I realize that RFC3261 says do not modify the From header, however, my > > question is this; > > > > When making SIP-to-SIP calls between two subscribers on the same SIP > > proxy, how can one implement Caller ID Block when RPID is not usually > > honored by SIP UAs - since this is really for PSTN gateways? > > > > It seems that altering the From header in this case is the only > > option. Have I missed something? > > > > Regards, > > Paul > > > > On Wed, 23 Feb 2005 12:00:14 +0100, Klaus Darilion > > <klaus.mailinglists(a)pernau.at> wrote: > > > Hi Mik! > > > > > > Do not modify the From: or To: URI - this is not allowed (RFC 3261) and > > > will cause problems. > > > > > > regards, > > > klaus > > > > > > Mik Cheez wrote: > > > > Since I'm able to modify the URI for relaying, I would assume there > > > > would be a way to modify the FROM_URI as well. Has anyone managed to do > > > > this with SER acting as a proxy? Any other possible avenues to modify > > > > the CALLID? > > > > > > > > Best regards > > > > > > > > _______________________________________________ > > > > Serusers mailing list > > > > serusers(a)lists.iptel.org > > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > > _______________________________________________ > > > Serusers mailing list > > > serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

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[Serusers] nat, symmetric /asymmetric UA....clarification needed
by Iqbal 27 Feb '05

27 Feb '05

Hi Just trying to debug why some calls work and other dont, I have 2 setups, in both the UA is xlite , and is natted in both cases...however the routers differ in each case. Now I am running mediaproxy, when I debug ser, in one setups the contact header which is displayed using xlog, shows the external IP address of the router, in the other it does not, this happens after I have passed the fix_contact setting in ser.cfg. The one where it is fixed , allows me to listen to voicemail prompt from asterisk without any problems, the other...no sound. The clarification I need is this.. 1. Mediaproxy solves symmetric NAT problems with ease..Y/N ? 2. For asymmetric stuff you need to add the clients to sip/rtp-asymmetric files Y/N ? 3 If Yes the the client in this case is xlite...which is not really the asymmetric part, its the router in the middle is it not ? 4. If I place slite in the file, will it not then affect the symmetric xlite also. Now if all this fails, can I on the xlite...which seems to be behind the asymmetric setup, use STUN server, if yes, then the mediaproxy I guess will not get called, or should it.....or is it better to map the port/pathsin the router itself. As a last question, if I wanted to determine what type of NATting a device did, or I needed to debug a customers setup, whats the easiest way tks Iqbal

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[Serusers] [Serdev] LCR module with gateway prefix
by Juha Heinanen 27 Feb '05

27 Feb '05

Raymond Chen writes: > Anyway to put the PSTN gateway prefix into gw table? no. gw table contains properties of gateways and routing info is in lcr table. -- juha

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[Serusers] LCR module with gateway prefix
by Raymond Chen 27 Feb '05

27 Feb '05

Dear Juha, Anyway to put the PSTN gateway prefix into gw table? Thanks Raymond

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[Serusers] bad udp packet? why!
by jiangjp 26 Feb '05

26 Feb '05

I installed the ser in Redhat 9.0. I use ngrep to capture the packet.Sometime I got the bad UDP packet send from ser. The following is an example(210.192.101.193 is the ser installed), before MESSAGE sip there are illeagal chars (maybe due to the udp size limit?): U 210.192.101.193:80 -> 218.25.90.134:31500 19036@0:1480 .P{.....MESSAGE sip:lumy1314@218.25.90.134:31500 SIP/2.0..Call-ID: 0f9 ffc6b4dbc35dd8cf153975330c83e@210.192.101.20..CSeq: 1 MESSAGE..From: " miphone" <sip:multichat@sip.miphone.cn:20000;transport=udp>;tag=160203 74..To: <sip:lumy1314@dododoo.com>..Via: SIP/2.0/UDP 210.192.101.193:8 0;branch=z9hG4bKacb8.7ed584b7.0..Via: SIP/2.0/UDP 210.192.101.20:20000 ;branch=z9hG4bK42e5063b387e084e01f8119c4491d60e..Max-Forwards: 69..Con tent-Type: text/x-miphone-multichat..Content-Length: 995..P-hint: usrl oc applied....<?xml version="1.0" encoding="UTF-8"?>.<miphone>.<comman d type="add user">.<from>sip:server</from>.<to>lumy1314(a)dododoo.com</t o>.<commandcontent>.<roomid>38563504</roomid>.<sipaddr>hxl(a)email.com.c n</sipaddr>.<name>tuhxl-1</name>.<sex>...</sex>.<age>4</age>.<imageid> 99999</imageid><location></location>.<userauthority>2</userauthority>. <userspeakstatus>NO</userspeakstatus>.<uservideostatus>NO</uservideost atus>.<audioaddr>61.149.14.202</audioaddr>.<audioport>28904</audioport >.<videoaddr>61.149.14.202</videoaddr>.<videoport>28914</videoport>.<n attype>1</nattype>.<firewalladdr></firewalladdr>.<localvideoaddr>61.14 9.14.202</localvideoaddr>.<localvideoport>28914</localvideoport>.<loca laudioaddr>61.149.14.202</localaudioaddr>.<localaudioport>28904</local audioport>.<nowuserlist>.<user>hxl(a)email.com.cn</user>.<user>tuhxl@mip hone.com.tv</user>.<user>bug(a)miphone.com.tv</user>.<user>lumy1314@dodo doo.com</user>.<user>czl(a)dododoo.com</user>.</nowuserlist>.</commandco ntent>.</c

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