- sr-users - kamailio.org

[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

2 1

[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

1 0

[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

2 1

[Serusers] lookup not finding local users
by Glenn Dalgliesh 26 Feb '05

26 Feb '05

I don't 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. 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.... ==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"); }; }

1 0

[Serusers] ser 0.9 startup problems
by Glenn Dalgliesh 26 Feb '05

26 Feb '05

I am getting the following errors when starting ser. I am sure it is something in my config but I can't track it down. This was a fresh install(no previous ver of ser on this machine.) I have include below the first section on my config. Just let me know if any one need the complete config Thanks Feb 25 18:54:11 baltodb ser[32594]: Maxfwd module- initializing Feb 25 18:54:11 baltodb ser[32594]: INFO: udp_init: SO_RCVBUF is initially 110592 Feb 25 18:54:11 baltodb ser[32594]: INFO: udp_init: SO_RCVBUF is finally 262142 Feb 25 18:54:11 baltodb ser[32594]: INFO: udp_init: SO_RCVBUF is initially 110592 Feb 25 18:54:11 baltodb ser[32594]: INFO: udp_init: SO_RCVBUF is finally 262142 Feb 25 18:54:11 baltodb ser[32604]: INFO: fifo process starting: 32604 Feb 25 18:54:11 baltodb ser[32604]: SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... Feb 25 18:54:11 baltodb ser[32604]: WARNING: no fifo_db_url given - fifo DB commands disabled! Feb 25 18:54:11 baltodb ser[32711]: new_connection(): Too many connections Feb 25 18:54:11 baltodb ser[32711]: db_init(): Could not create a connection Feb 25 18:54:11 baltodb ser[32711]: ERROR: uridb_db_bind: unable to connect to the database Feb 25 18:54:11 baltodb ser[32711]: init_mod_child(): Error while initializing module uri_db Feb 25 18:54:11 baltodb ser[32711]: init_children failed Feb 25 18:54:11 baltodb ser[32604]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32607]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32613]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32616]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32619]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32620]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32623]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32626]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32629]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32632]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32635]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32641]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32644]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32610]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32636]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32647]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32650]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32653]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32655]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32664]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32662]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32667]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32670]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32656]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32673]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32675]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32683]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32684]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32676]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32685]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32686]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32687]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32688]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32695]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32697]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32698]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32699]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32700]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32701]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32709]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32702]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32710]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32724]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32725]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32712]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32713]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32714]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32722]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32723]: INFO: signal 15 received Feb 25 18:54:11 baltodb ser[32726]: INFO: signal 15 received Feb 25 18:54:12 baltodb ser[32594]: BUG: qm_free: freeing already freed pointer, first free: ms_msg_list.c: msg_list_free(162) - aborting # # $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.popvox.com 2.6.10-1.760_FC3 #1 Wed Feb 2 00:14:23 EST 2005 i686 i686 i386 GNU/Linux # date: Thu Feb 24 15:51:05 EST 2005 # # ----------- global configuration parameters ------------------------ debug=3 fork=yes port=5060 log_stderror=no memlog=5 mhomed=yes fifo="/tmp/ser_fifo" alias=popvox.com # uncomment to override config values for test /* debug=3 # debug level (cmd line: -ddd) fork=no port=5068 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", "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@popvox.com") # -- enum params -- # modparam("enum", "domain_suffix", "e164.arpa.") # ------------------------- request routing logic ------------------- # main routing logic

2 2

[Serusers] METHOD<null>
by Iqbal 26 Feb '05

26 Feb '05

Hi This doesnt seem to be a standard SIP method, is there any reason I would be getting this in my logs, it always seems to come from the called party..in this case being xlite Iqbal

3 3

[Serusers] contact header fix
by Iqbal 26 Feb '05

26 Feb '05

Hi When I use fix_contact with the mediaproxy module, should not the contact which is displayed using xlog namely %ct change Iqbal

2 2

[Serusers] UDP packet is bad?
by xiahong 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

1 0

[Serusers] forwarding to pstn
by harry gaillac 25 Feb '05

25 Feb '05

Hi all, I set up ser + asterisk. Asterisk is registred as gateway(a)home.net and listen port 5070 (same box). two hard phones are registred as bob(a)home.net and alice(a)home.net . from alice or bob i want to call 01013 to PSTN without success. ser lookup location and aliases and so reply not found thought i set forward ? Thanks for your help Harry # ----------- global configuration parameters ------------------------ debug=5 # 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) listen=192.168.0.1 alias="home.net" #alias="192.168.0.1" port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # 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/group.so" loadmodule "/usr/lib/ser/modules/uri.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) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("15")) { 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) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("home.net", "subscriber")) { www_challenge("home.net", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB or ALIASES if (!lookup("location") | !lookup("aliases")) { sl_send_reply("404", "Not Found"); break; }; } else { t_relay(); }; # 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(); }; # forward to PSTN if (uri=~"sip:01013@home.net") { if (method=="INVITE") { route(1); break; }; }; } route[1] { # forward to PSTN if (uri=~"sip:01013@home.net") { if (method=="INVITE") { forward(192.168.0.1, 5070); break; }; }; } Découvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com/ Découvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com/

3 2

[Serusers] include_route
by Iqbal 25 Feb '05

25 Feb '05

Hi In my ser.cfg I seem to have lots of segments which are repeated in each route, is there anyway...or would/can it be included in a later realease a method of including these i.e include_segment[1] { Check if authorised } route [1] { include_segment(1) rest of stuff } Iqbal

2 1

[Serusers] find_export: <append_hf> not found - SOLVED IT
by Paulo Vicentini 25 Feb '05

25 Feb '05

Hi Lucas, Would mind sending your ser.cfg file, i am facing the same 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: # # $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

1 0

[Serusers] SER Authentication Configuration FAILLER
by Brahim 25 Feb '05

25 Feb '05

I am using ser-0.9.0 and freeradius-0.9.1, i install free radius succefull but when i try to recompile ser the auth_radius.so module doest not create and i get the following errors. /***************************************************************************************************************************************************/ make[1]: Entering directory `/usr/ser-0.9.0/modules/group_radius' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=i686 -DNAME='"ser"' -DVERSION='"0.9.0"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.3"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -I/usr/local/include -c group.c -o group.o In file included from ../../parser/parse_uri.h:38, from group.c:42: ../../parser/msg_parser.h: In function `get_body': ../../parser/msg_parser.h:314: warning: comparison between signed and unsigned ../../parser/msg_parser.h:316: warning: comparison between signed and unsigned group.c: In function `radius_is_user_in': group.c:194: warning: passing arg 2 of `rc_avpair_add' makes integer from pointer without a cast group.c:194: warning: passing arg 3 of `rc_avpair_add' makes pointer from integer without a cast group.c:194: warning: passing arg 4 of `rc_avpair_add' makes integer from pointer without a cast group.c:194: error: too many arguments to function `rc_avpair_add' group.c:203: warning: passing arg 2 of `rc_avpair_add' makes integer from pointer without a cast group.c:203: warning: passing arg 3 of `rc_avpair_add' makes pointer from integer without a cast group.c:203: warning: passing arg 4 of `rc_avpair_add' makes integer from pointer without a cast group.c:203: error: too many arguments to function `rc_avpair_add' group.c:209: warning: passing arg 2 of `rc_avpair_add' makes integer from pointer without a cast group.c:209: warning: passing arg 3 of `rc_avpair_add' makes pointer from integer without a cast group.c:209: warning: passing arg 4 of `rc_avpair_add' makes integer from pointer without a cast group.c:209: error: too many arguments to function `rc_avpair_add' group.c:215: warning: passing arg 1 of `rc_auth' makes integer from pointer without a cast group.c:215: warning: passing arg 3 of `rc_auth' from incompatible pointer type group.c:215: warning: passing arg 4 of `rc_auth' from incompatible pointer type group.c:215: error: too many arguments to function `rc_auth' make[1]: *** [group.o] Error 1 /***************************************************************************************************************************************************/ Any Help.

2 1

[Serusers] SER+Asterisk on high loads
by Alexey N. Kovyrin ＠ Home 25 Feb '05

25 Feb '05

Hello, All! There is some questions about subj. With what loads SER+Asterisk can be used? I need to know, what number of simultaneous registrations and simultaneous calls can handle this conplex? It'll be very good, if someone can say his success story about SER+Asterisk. Now, we're planning 1000-10000 clients... can SER handle this number of registrations? And can SER+Asterisk (assume that almost all users are using NAT) can handle calls of 0.1-1% of above clients base? Great thanks to all for answers. -- /Scoundrel

4 7

[Serusers] ;lr=on> versus ;lr> -- which is RFC3261 compliant?
by Java Rockx 25 Feb '05

25 Feb '05

Hi all. We have a partner with a Sonus box that we use for PSTN termination. Their Sonus box produces suspect Record-Route headers. Can anyone tell me if it is compliant with RFC3261? A sample header that I receive looks like this: Record-Route: <sip:xxx.xxx.xxx.xxx:5060;lr> And I discovered that Asterisk-1.0.2 seems to not properly handle these messages so it does strict routing rather than loose routing. Greg greger(a)teigre.com was kind enough to point me to RFC3261 Section 19.1.1 which seems to indicate that ;lr> should be ;lr=on> for complance. Is this correct? Regards, Paul

3 3

[Serusers] SER, rtpproxy and SIPclient
by Mohammad Khan 25 Feb '05

25 Feb '05

My SER is running behind NAT with rtpproxy. I was able to send and receive message via Windows Messenger. Both clients were outside my network and they were also behind NAT. Problem is, I can't communicate with a client (outside my network) from my internal network. Here is my iptables rules -A PREROUTING -p tcp -d 66.92.x.y --dport 5060 -j DNAT --to-destination 192.168.71.2 -A PREROUTING -p udp -d 66.92.x.y --dport 5060 -j DNAT --to-destination 192.168.71.2 -A POSTROUTING -s 192.168.71.0/24 -d 192.168.71.2 -p tcp --dport 5060 -j SNAT --to 192.168.71.1 -A POSTROUTING -s 192.168.71.0/24 -d 192.168.71.2 -p udp --dport 5060 -j SNAT --to 192.168.71.1 -A POSTROUTING -s 192.168.71.0/24 -o eth0 -j SNAT --to-source 66.92.x.y my rtpproxy is running without any argument. Would anybody please give me hint, how to overcome this problem. Thanks, Mohammad

1 0

[Serusers] help a newbie !!!
by Atif Rasheed 25 Feb '05

25 Feb '05

hello there, I am new to SER, just want to configure it with two NIC interfaces (for global network. calls from external network (global network) should be routed to a sip-endpoint in internal network. and I need rtp proxied with it. I have tried but rtpproxy is crashing or SER is crashing. can some one please tell me the stable version of SER or rtpproxy to use and can some one please send me his configuration file working in a similar schema. *sip-endpoint in internal network is not registered on the SER. thanks Atif

1 0

RE: [Serusers] Turn 3xx into new Invite
by Paul Belanger 25 Feb '05

25 Feb '05

Just got a reply from a fellow SER user to add append_branch() fuction, but I still cannot see the INVITE from my SER. onreply_route[1] { xlog("L_NOTICE", "Reply status %rs: Reason %rr\n"); if (t_check_status("302")) { xlog("L_NOTICE", "We got a 302!!!!!\n"); ## Used to send INVITE message? ## append_branch(); }; } -----Original Message----- From: Martin Koenig [mailto:martin.koenig@toplink-plannet.de] Sent: Tuesday, February 22, 2005 11:33 AM To: Paul Belanger Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Turn 3xx into new Invite Hello, see: http://lists.iptel.org/pipermail/serusers/2004-December/014107.html looks like Juha Heinänen has a patch for this available somewhere, maybe he would like to share this? Regards, Martin Paul Belanger schrieb: >Ya, I have looked at that one, but I can see how it sends out the >INVITE. > >-----Original Message----- >From: Martin Koenig [mailto:martin.koenig@toplink-plannet.de] >Sent: Tuesday, February 22, 2005 11:09 AM >To: Paul Belanger >Cc: serusers(a)lists.iptel.org >Subject: Re: [Serusers] Turn 3xx into new Invite > > >Hello, > >there is this one so far: > >http://lists.iptel.org/pipermail/serusers/2004-August/010977.html > >Didn't really help... > >Regards, >Martin > >Paul Belanger schrieb: > > > >>Rumor has it yes, but I have not been successful. I posted a question >>last week, and was told to search the archives. I did, but not much >>came up! >> >>http://www.google.ca/search?hl=en&q=302+site%3Amail.iptel.org&btnG=Goo >>g >>le+Search&meta= >> >>I think I'm on the right track, but don't have time today to play with >>it. Here is what I got so far: >> >># >># $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) >> >>debug=3 >>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 ---------------------------------- >> >># 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/xlog.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") >> >># -- rr params -- >># add value to ;lr param to make some broken UAs happy modparam("rr", >>"enable_full_lr", 1) >> >>modparam("tm", "fr_inv_timer", 15 ) >>modparam("tm", "fr_timer", 10 ) >> >># ------------------------- 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) { >> >> if (method=="REGISTER") { >> save("location"); >> break; >> }; >> }; >> >> >> if (method=="INVITE") { >> rewritehostport("216.143.130.70:5060"); >> t_on_reply("1"); >> t_on_failure("1"); >> t_relay(); >> break; >> } >> >>} >> >>onreply_route[1] { >> xlog("L_NOTICE", "Reply status %rs: Reason %rr\n"); >> if (t_check_status("302")) { >> xlog("L_NOTICE", "We got a 302!!!!!\n"); >> ## ADD CODE TO SEND OUT INVITE!! ### >> }; >>} >> >>failure_route[1] { >> xlog("L_NOTICE", "Failure status %rs: Reason %rr\n"); } >> >>Let me know if you find any information. >> >>Paul >> >>-----Original Message----- >>From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] >>On >>Behalf Of Martin Koenig >>Sent: Tuesday, February 22, 2005 11:00 AM >>To: serusers(a)lists.iptel.org >>Subject: [Serusers] Turn 3xx into new Invite >> >> >>Hello, >> >>is it possible to take 3xx-responses in a reply_route and turn them >>into >>a new INVITE on Ser using append_branch or simliar? >> >>Regards, >>Martin >> >>_______________________________________________ >>Serusers mailing list >>serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> >> > > >

5 5

[Serusers] aliases with trigger problem
by braincrew.com 25 Feb '05

25 Feb '05

Hi guys, I tried to make aliases w/ trigger but something goes wrong. Trigger is made on insert in the database. If user logins with username 123 alias 0123 is created. But if I call 0123 I get "404 Not found" any idea? P.S. in ser.cfg I have : if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("193.193.162.71", "subscriber")) { www_challenge("193.193.162.71", "0"); break; }; save("location"); lookup("aliases"); Thanks!

2 3

[Serusers] Re: Serusers Digest, Vol 22, Issue 35
by Brahim 25 Feb '05

25 Feb '05

I try to forward i call to asterisk via SER, the call must be executed in a context named [XXXXX], how can i do this.

2 1

Re: [Serusers] Questions on using ser.cfg version 0.9.0 for call forward/busy/noanswer
by Charles Wang 25 Feb '05

25 Feb '05

Dear Paul: UA1(1011) ==> SER ==> UA(9012) ==> busy forward to ==> CISCO Trunking. Using Route[2] failure_route[1] Using route[6] I find it go to route[6], but failed for a little bit seconds. Please help me to find what's it wrong? I add the log messages and ngrep result below: Log from /var/log/messages: ------------------------------------------------------------------------------------------------------------------- ser[27626]: PDT:prefix2domain: no prefix found in [9012] ser[27626]: Time:[Fri 2005] Method:<INVITE> r-uri:<9012(a)ser.xxx.net.tw> ser[27626]: IP:<61.229.4.212> From:<sip:1011@ser.xxx.net.tw> To:<sip:9012@ser.xxx.net.tw> <sip:1011@61.229.4.212:5060> ser[27626]: SER: a INT user ser[27626]: SER: BLIND CALL FORWARDING ser[27626]: SER: Look aliases ser[27626]: SER: Look location ser[27626]: SER isflagset (sip) ser[27626]: SER: Look aliases ser[27626]: SER: Look location ser[27626]: SER isflagset (sip) ser[27626]: SER: SIP Call On-Net section route(2) ser[27627]: PDT:prefix2domain: no prefix found in [9012] ser[27627]: Time:[Fri 2005] Method:<INVITE> r-uri:<9012(a)ser.xxx.net.tw> ser[27627]: IP:<61.229.4.212> From:<sip:1011@ser.xxx.net.tw> To:<sip:9012@ser.xxx.net.tw> <sip:1011@61.229.4.212:5060> ser[27627]: SER: a INT user ser[27627]: SER: BLIND CALL FORWARDING ser[27627]: SER: Look aliases ser[27627]: SER: Look location ser[27627]: SER isflagset (sip) ser[27627]: SER: Look aliases ser[27627]: SER: Look location ser[27627]: SER isflagset (sip) ser[27627]: SER: SIP Call On-Net section route(2) ser proxydispatcher[27685]: command request 40088117-5908-46A9-8D9B-E872430672FE(a)10.18.1.70 61.229.4.212:8000:audio 61.229.4.212 ser.xxx.net.tw local 61.229.4.212 remote X-PRO=20build=201082 info=from:1011@ser.xxx.net.tw,to:9012@ser.xxx.net.tw,fromtag:903024600,totag: ser proxydispatcher[27685]: domain ser.xxx.net.tw doesn't define any mediaproxy. ser proxydispatcher[27685]: will use default mediaproxy for this call. ser mediaproxy[1087]: command request 40088117-5908-46A9-8D9B-E872430672FE(a)10.18.1.70 61.229.4.212:8000:audio 61.229.4.212 ser.xxx.net.tw local 61.229.4.212 remote X-PRO=20build=201082 info=from:1011@ser.xxx.net.tw,to:9012@ser.xxx.net.tw,fromtag:903024600,totag:,dispatcher ser mediaproxy[1087]: session 40088117-5908-46A9-8D9B-E872430672FE(a)10.18.1.70: started. listening on xxx.xxx.190.248:35120 ser mediaproxy[1087]: command execution time: 6.08 ms ser proxydispatcher[27685]: forwarding to mediaproxy on /var/run/mediaproxy.sock: got: 'xxx.xxx.190.248 35120' ser proxydispatcher[27685]: command execution time: 96.11 ms ser[27628]: SER: Failure Route section failure_route(1) ser[27628]: SER: fork to fwdbusy ser[27628]: SER: Busy Failure and Jump to route(6) ser[27628]: SER: Domestic Call Off-Net section route(6) ser proxydispatcher[27686]: command request 40088117-5908-46A9-8D9B-E872430672FE(a)10.18.1.70 61.229.4.212:8000:audio 61.229.4.212 ser.xxx.net.tw local xxx.xxx.190.243 remote X-PRO=20build=201082 info=from:1011@ser.xxx.net.tw,to:9012@ser.xxx.net.tw,fromtag:903024600,totag: Feb 25 15:56:46 ser proxydispatcher[27686]: domain ser.xxx.net.tw doesn't define any mediaproxy. Feb 25 15:56:46 ser proxydispatcher[27686]: will use default mediaproxy for this call. Feb 25 15:56:46 ser mediaproxy[1087]: command request 40088117-5908-46A9-8D9B-E872430672FE(a)10.18.1.70 61.229.4.212:8000:audio 61.229.4.212 ser.xxx.net.tw local xxx.xxx.190.243 remote X-PRO=20build=201082 info=from:1011@ser.xxx.net.tw,to:9012@ser.xxx.net.tw,fromtag:903024600,totag:,dispatcher Feb 25 15:56:46 ser mediaproxy[1087]: command execution time: 0.35 ms Feb 25 15:56:46 ser proxydispatcher[27686]: forwarding to mediaproxy on /var/run/mediaproxy.sock: got: 'xxx.xxx.190.248 35120' Feb 25 15:56:46 ser proxydispatcher[27686]: command execution time: 3763.49 ms Feb 25 15:57:05 ser[27627]: PDT:prefix2domain: no prefix found in [9012] Feb 25 15:57:05 ser[27627]: SER: SIP Call On-Net section route(2) [root@ser root]# Log from ngrep: -------------------------------------------------------------------------------------------------------- [root@ser root]# ngrep -d eth0 -W byline port 5060 interface: eth0 (xxx.xxx.190.240/255.255.255.240) filter: ip and ( port 5060 ) # U xxx.xxx.190.248:5060 -> 61.229.4.212:5060 .... # U xxx.xxx.190.248:5060 -> 61.217.225.150:1720 .... # U xxx.xxx.190.248:5060 -> 61.229.4.212:35375 .... # U 61.229.4.212:35971 -> xxx.xxx.190.248:5060 INVITE sip:9012@ser.xxx.net.tw SIP/2.0. Via: SIP/2.0/UDP 61.229.4.212:5060;rport;branch=z9hG4bK2BD8C0F5D91D44FCBC6A145F42A8D0EA. From: 1011 <sip:1011@ser.xxx.net.tw>;tag=1163705917. To: <sip:9012@ser.xxx.net.tw>. Contact: <sip:1011@61.229.4.212:5060>. Call-ID: 4891EE7A-B7E2-40EA-B8E0-6CB7F5045F9F(a)10.18.1.70. CSeq: 61468 INVITE. Max-Forwards: 70. Content-Type: application/sdp. User-Agent: X-PRO build 1082. Content-Length: 320. . v=0. o=1011 133082262 133082262 IN IP4 61.229.4.212. s=X-PRO. c=IN IP4 61.229.4.212. t=0 0. m=audio 8000 RTP/AVP 0 8 3 18 98 97 101. a=rtpmap:0 pcmu/8000. a=rtpmap:8 pcma/8000. a=rtpmap:3 gsm/8000. a=rtpmap:18 G729/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:97 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U xxx.xxx.190.248:5060 -> 61.229.4.212:35971 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 61.229.4.212:5060;rport=35971;branch=z9hG4bK2BD8C0F5D91D44FCBC6A145F42A8D0EA. From: 1011 <sip:1011@ser.xxx.net.tw>;tag=1163705917. To: <sip:9012@ser.xxx.net.tw>;tag=67771a809cdfb71129a699a517fbb1f0.3a1e. Call-ID: 4891EE7A-B7E2-40EA-B8E0-6CB7F5045F9F(a)10.18.1.70. CSeq: 61468 INVITE. Proxy-Authenticate: Digest realm="ser.xxx.net.tw", nonce="421edc7e7c26aa218feb652ab31161d4f73a3fe6". Server: Sip EXpress router (0.10.99-dev0 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.190.248:5060 "Noisy feedback tells: pid=27628 req_src_ip=61.229.4.212 req_src_port=35971 in_uri=sip:9012@ser.xxx.net.tw out_uri=sip:9012@61.229.4.212:35375 via_cnt==1". . # U 61.229.4.212:35971 -> xxx.xxx.190.248:5060 ACK sip:9012@ser.xxx.net.tw SIP/2.0. Via: SIP/2.0/UDP 61.229.4.212:5060;rport;branch=z9hG4bK2BD8C0F5D91D44FCBC6A145F42A8D0EA. From: 1011 <sip:1011@ser.xxx.net.tw>;tag=1163705917. To: <sip:9012@ser.xxx.net.tw>;tag=67771a809cdfb71129a699a517fbb1f0.3a1e. Contact: <sip:1011@61.229.4.212:5060>. Call-ID: 4891EE7A-B7E2-40EA-B8E0-6CB7F5045F9F(a)10.18.1.70. CSeq: 61468 ACK. Max-Forwards: 70. Content-Length: 0. . # U 61.229.4.212:35971 -> xxx.xxx.190.248:5060 INVITE sip:9012@ser.xxx.net.tw SIP/2.0. Via: SIP/2.0/UDP 61.229.4.212:5060;rport;branch=z9hG4bKD4F73B7C0DFA47BD8441B0B8A028E621. From: 1011 <sip:1011@ser.xxx.net.tw>;tag=1163705917. To: <sip:9012@ser.xxx.net.tw>. Contact: <sip:1011@61.229.4.212:5060>. Call-ID: 4891EE7A-B7E2-40EA-B8E0-6CB7F5045F9F(a)10.18.1.70. CSeq: 61469 INVITE. Proxy-Authorization: Digest username="1011",realm="ser.xxx.net.tw",nonce="421edc7e7c26aa218feb652ab31161d4f73a3fe6",response="67bf9eeee503891ded4971a4589719bd",uri="sip:9012@ser.xxx.net.tw". Max-Forwards: 70. Content-Type: application/sdp. User-Agent: X-PRO build 1082. Content-Length: 320. . v=0. o=1011 133083073 133083073 IN IP4 61.229.4.212. s=X-PRO. c=IN IP4 61.229.4.212. t=0 0. m=audio 8000 RTP/AVP 0 8 3 18 98 97 101. a=rtpmap:0 pcmu/8000. a=rtpmap:8 pcma/8000. a=rtpmap:3 gsm/8000. a=rtpmap:18 G729/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:97 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 61.229.4.212:35971 -> xxx.xxx.190.248:5060 INVITE sip:9012@ser.xxx.net.tw SIP/2.0. Via: SIP/2.0/UDP 61.229.4.212:5060;rport;branch=z9hG4bKD4F73B7C0DFA47BD8441B0B8A028E621. From: 1011 <sip:1011@ser.xxx.net.tw>;tag=1163705917. To: <sip:9012@ser.xxx.net.tw>. Contact: <sip:1011@61.229.4.212:5060>. Call-ID: 4891EE7A-B7E2-40EA-B8E0-6CB7F5045F9F(a)10.18.1.70. CSeq: 61469 INVITE. Proxy-Authorization: Digest username="1011",realm="ser.xxx.net.tw",nonce="421edc7e7c26aa218feb652ab31161d4f73a3fe6",response="67bf9eeee503891ded4971a4589719bd",uri="sip:9012@ser.xxx.net.tw". Max-Forwards: 70. Content-Type: application/sdp. User-Agent: X-PRO build 1082. Content-Length: 320. . v=0. o=1011 133084745 133084745 IN IP4 61.229.4.212. s=X-PRO. c=IN IP4 61.229.4.212. t=0 0. m=audio 8000 RTP/AVP 0 8 3 18 98 97 101. a=rtpmap:0 pcmu/8000. a=rtpmap:8 pcma/8000. a=rtpmap:3 gsm/8000. a=rtpmap:18 G729/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:97 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U xxx.xxx.190.248:5060 -> 61.229.4.212:35971 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 61.229.4.212:5060;rport=35971;branch=z9hG4bKD4F73B7C0DFA47BD8441B0B8A028E621. From: 1011 <sip:1011@ser.xxx.net.tw>;tag=1163705917. To: <sip:9012@ser.xxx.net.tw>. Call-ID: 4891EE7A-B7E2-40EA-B8E0-6CB7F5045F9F(a)10.18.1.70. CSeq: 61469 INVITE. Server: Sip EXpress router (0.10.99-dev0 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.190.248:5060 "Noisy feedback tells: pid=27627 req_src_ip=61.229.4.212 req_src_port=35971 in_uri=sip:9012@ser.xxx.net.tw out_uri=sip:9012@61.229.4.212:35375 via_cnt==1". . # U xxx.xxx.190.248:5060 -> 61.229.4.212:35375 INVITE sip:9012@61.229.4.212:35375 SIP/2.0. Record-Route: <sip:xxx.xxx.190.248;ftag=1163705917;lr=on>. Via: SIP/2.0/UDP xxx.xxx.190.248;branch=z9hG4bK467d.fd6a4256.0. Via: SIP/2.0/UDP 61.229.4.212:5060;rport=35971;branch=z9hG4bKD4F73B7C0DFA47BD8441B0B8A028E621. From: 1011 <sip:1011@ser.xxx.net.tw>;tag=1163705917. To: <sip:9012@ser.xxx.net.tw>. Contact: <sip:1011@61.229.4.212:5060>. Call-ID: 4891EE7A-B7E2-40EA-B8E0-6CB7F5045F9F(a)10.18.1.70. CSeq: 61469 INVITE. Proxy-Authorization: Digest username="1011",realm="ser.xxx.net.tw",nonce="421edc7e7c26aa218feb652ab31161d4f73a3fe6",response="67bf9eeee503891ded4971a4589719bd",uri="sip:9012@ser.xxx.net.tw". Max-Forwards: 16. Content-Type: application/sdp. User-Agent: X-PRO build 1082. Content-Length: 323. . v=0. o=1011 133084745 133084745 IN IP4 61.229.4.212. s=X-PRO. c=IN IP4 xxx.xxx.190.248. t=0 0. m=audio 35122 RTP/AVP 0 8 3 18 98 97 101. a=rtpmap:0 pcmu/8000. a=rtpmap:8 pcma/8000. a=rtpmap:3 gsm/8000. a=rtpmap:18 G729/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:97 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U xxx.xxx.190.248:5060 -> 61.229.4.212:35375 INVITE sip:9012@61.229.4.212:35375 SIP/2.0. Record-Route: <sip:xxx.xxx.190.248;ftag=1163705917;lr=on>. Via: SIP/2.0/UDP xxx.xxx.190.248;branch=z9hG4bK467d.fd6a4256.0. Via: SIP/2.0/UDP 61.229.4.212:5060;rport=35971;branch=z9hG4bKD4F73B7C0DFA47BD8441B0B8A028E621. From: 1011 <sip:1011@ser.xxx.net.tw>;tag=1163705917. To: <sip:9012@ser.xxx.net.tw>. Contact: <sip:1011@61.229.4.212:5060>. Call-ID: 4891EE7A-B7E2-40EA-B8E0-6CB7F5045F9F(a)10.18.1.70. CSeq: 61469 INVITE. Proxy-Authorization: Digest username="1011",realm="ser.xxx.net.tw",nonce="421edc7e7c26aa218feb652ab31161d4f73a3fe6",response="67bf9eeee503891ded4971a4589719bd",uri="sip:9012@ser.xxx.net.tw". Max-Forwards: 16. Content-Type: application/sdp. User-Agent: X-PRO build 1082. Content-Length: 323. . v=0. o=1011 133084745 133084745 IN IP4 61.229.4.212. s=X-PRO. c=IN IP4 xxx.xxx.190.248. t=0 0. m=audio 35122 RTP/AVP 0 8 3 18 98 97 101. a=rtpmap:0 pcmu/8000. a=rtpmap:8 pcma/8000. a=rtpmap:3 gsm/8000. a=rtpmap:18 G729/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:97 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 61.229.4.212:35375 -> xxx.xxx.190.248:5060 SIP/2.0 486 Busy Here. Via: SIP/2.0/UDP xxx.xxx.190.248;branch=z9hG4bK467d.fd6a4256.0. Via: SIP/2.0/UDP 61.229.4.212:5060;rport=35971;branch=z9hG4bKD4F73B7C0DFA47BD8441B0B8A028E621. Record-Route: <sip:xxx.xxx.190.248;ftag=1163705917;lr=on>. Call-ID: 4891EE7A-B7E2-40EA-B8E0-6CB7F5045F9F(a)10.18.1.70. CSeq: 61469 INVITE. From: 1011 <sip:1011@ser.xxx.net.tw>;tag=1163705917. To: <sip:9012@ser.xxx.net.tw>;tag=dk09jcgubVdV7nmT. Contact: <sip:9012@10.18.1.102:1718>. Content-Length: 0. . # U xxx.xxx.190.248:5060 -> 61.229.4.212:35375 ACK sip:9012@61.229.4.212:35375 SIP/2.0. Via: SIP/2.0/UDP xxx.xxx.190.248;branch=z9hG4bK467d.fd6a4256.0. From: 1011 <sip:1011@ser.xxx.net.tw>;tag=1163705917. Call-ID: 4891EE7A-B7E2-40EA-B8E0-6CB7F5045F9F(a)10.18.1.70. To: <sip:9012@ser.xxx.net.tw>;tag=dk09jcgubVdV7nmT. CSeq: 61469 ACK. User-Agent: Sip EXpress router(0.10.99-dev0 (i386/linux)). Content-Length: 0. . # U 61.229.4.212:35375 -> xxx.xxx.190.248:5060 SIP/2.0 486 Busy Here. Via: SIP/2.0/UDP xxx.xxx.190.248;branch=z9hG4bK467d.fd6a4256.0. Via: SIP/2.0/UDP 61.229.4.212:5060;rport=35971;branch=z9hG4bKD4F73B7C0DFA47BD8441B0B8A028E621. Record-Route: <sip:xxx.xxx.190.248;ftag=1163705917;lr=on>. Call-ID: 4891EE7A-B7E2-40EA-B8E0-6CB7F5045F9F(a)10.18.1.70. CSeq: 61469 INVITE. From: 1011 <sip:1011@ser.xxx.net.tw>;tag=1163705917. To: <sip:9012@ser.xxx.net.tw>;tag=dk09jcgubVdV7nmT. Contact: <sip:9012@10.18.1.102:1718>. Content-Length: 0. . # U xxx.xxx.190.248:5060 -> 61.229.4.212:35375 ACK sip:9012@61.229.4.212:35375 SIP/2.0. Via: SIP/2.0/UDP xxx.xxx.190.248;branch=z9hG4bK467d.fd6a4256.0. From: 1011 <sip:1011@ser.xxx.net.tw>;tag=1163705917. Call-ID: 4891EE7A-B7E2-40EA-B8E0-6CB7F5045F9F(a)10.18.1.70. To: <sip:9012@ser.xxx.net.tw>;tag=dk09jcgubVdV7nmT. CSeq: 61469 ACK. User-Agent: Sip EXpress router(0.10.99-dev0 (i386/linux)). Content-Length: 0. . # U xxx.xxx.190.248:5060 -> 61.229.4.212:35971 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 61.229.4.212:5060;rport=35971;branch=z9hG4bKD4F73B7C0DFA47BD8441B0B8A028E621. From: 1011 <sip:1011@ser.xxx.net.tw>;tag=1163705917. To: <sip:9012@ser.xxx.net.tw>. Call-ID: 4891EE7A-B7E2-40EA-B8E0-6CB7F5045F9F(a)10.18.1.70. CSeq: 61469 INVITE. Server: Sip EXpress router (0.10.99-dev0 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.190.248:5060 "Noisy feedback tells: pid=27627 req_src_ip=61.229.4.212 req_src_port=35971 in_uri=sip:9012@ser.xxx.net.tw out_uri=sip:9012@61.229.4.212:35375 via_cnt==1". . # U 61.229.4.212:35375 -> xxx.xxx.190.248:5060 REGISTER sip:ser.xxx.net.tw SIP/2.0. Via: SIP/2.0/UDP 10.18.1.102:1718;branch=z9hG4bKRSezesZsx. Max-Forwards: 70. User-Agent: PHONE. From: "9012" <sip:9012@ser.xxx.net.tw>;tag=bD35hL0X5YQwOZZg. To: "9012" <sip:9012@ser.xxx.net.tw>. Call-ID: jLNZpAzRan2jYdn5(a)10.18.1.102. CSeq: 29076 REGISTER. Contact: <sip:9012@10.18.1.102:1718>. Expires: 60. Content-Length: 0. . # U xxx.xxx.190.248:5060 -> 61.229.4.212:35375 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 10.18.1.102:1718;branch=z9hG4bKRSezesZsx;rport=35375;received=61.229.4.212. From: "9012" <sip:9012@ser.xxx.net.tw>;tag=bD35hL0X5YQwOZZg. To: "9012" <sip:9012@ser.xxx.net.tw>;tag=67771a809cdfb71129a699a517fbb1f0.d163. Call-ID: jLNZpAzRan2jYdn5(a)10.18.1.102. CSeq: 29076 REGISTER. WWW-Authenticate: Digest realm="ser.xxx.net.tw", nonce="421edc879a81096d6cc9e10d88f1aa394f199b57". Server: Sip EXpress router (0.10.99-dev0 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.190.248:5060 "Noisy feedback tells: pid=27627 req_src_ip=61.229.4.212 req_src_port=35375 in_uri=sip:ser.xxx.net.tw out_uri=sip:ser.xxx.net.tw via_cnt==1". . # U 61.229.4.212:35375 -> xxx.xxx.190.248:5060 REGISTER sip:ser.xxx.net.tw SIP/2.0. Via: SIP/2.0/UDP 10.18.1.102:1718;branch=z9hG4bKzsFNXbhF2. Max-Forwards: 70. User-Agent: PHONE. From: "9012" <sip:9012@ser.xxx.net.tw>;tag=5GvyGNvy5GcfQrx1. To: "9012" <sip:9012@ser.xxx.net.tw>. Call-ID: jLNZpAzRan2jYdn5(a)10.18.1.102. CSeq: 29077 REGISTER. Contact: <sip:9012@10.18.1.102:1718>. Expires: 60. Authorization: Digest username="9012", realm="ser.xxx.net.tw", nonce="421edc879a81096d6cc9e10d88f1aa394f199b57", uri="sip:ser.xxx.net.tw", response="5c4a89bcb17c5bda756db998ee704fbe". Content-Length: 0. . # U xxx.xxx.190.248:5060 -> 61.229.4.212:35375 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.18.1.102:1718;branch=z9hG4bKzsFNXbhF2;rport=35375;received=61.229.4.212. From: "9012" <sip:9012@ser.xxx.net.tw>;tag=5GvyGNvy5GcfQrx1. To: "9012" <sip:9012@ser.xxx.net.tw>;tag=67771a809cdfb71129a699a517fbb1f0.2657. Call-ID: jLNZpAzRan2jYdn5(a)10.18.1.102. CSeq: 29077 REGISTER. P-NAT-Refresh: 15. Contact: <sip:9012@61.229.4.212:35375>;expires=60. Server: Sip EXpress router (0.10.99-dev0 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.190.248:5060 "Noisy feedback tells: pid=27628 req_src_ip=61.229.4.212 req_src_port=35375 in_uri=sip:ser.xxx.net.tw out_uri=sip:ser.xxx.net.tw via_cnt==1". . # U xxx.xxx.190.248:5060 -> 61.229.4.212:35971 SIP/2.0 408 Request Timeout. <================ DISCONNECT HERE ======================== Via: SIP/2.0/UDP 61.229.4.212:5060;rport=35971;branch=z9hG4bKD4F73B7C0DFA47BD8441B0B8A028E621. From: 1011 <sip:1011@ser.xxx.net.tw>;tag=1163705917. To: <sip:9012@ser.xxx.net.tw>;tag=5f573bbafd260ada15def89f1b1724a2-cff8. Call-ID: 4891EE7A-B7E2-40EA-B8E0-6CB7F5045F9F(a)10.18.1.70. CSeq: 61469 INVITE. Server: Sip EXpress router (0.10.99-dev0 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.190.248:5060 "Noisy feedback tells: pid=27630 req_src_ip=61.229.4.212 req_src_port=35971 in_uri=sip:9012@ser.xxx.net.tw out_uri=sip:9012@61.229.4.212:35375 via_cnt==0". . # U 61.229.4.212:35971 -> xxx.xxx.190.248:5060 ACK sip:9012@ser.xxx.net.tw SIP/2.0. Via: SIP/2.0/UDP 61.229.4.212:5060;rport;branch=z9hG4bKD4F73B7C0DFA47BD8441B0B8A028E621. From: 1011 <sip:1011@ser.xxx.net.tw>;tag=1163705917. To: <sip:9012@ser.xxx.net.tw>;tag=5f573bbafd260ada15def89f1b1724a2-cff8. Contact: <sip:1011@61.229.4.212:5060>. Call-ID: 4891EE7A-B7E2-40EA-B8E0-6CB7F5045F9F(a)10.18.1.70. CSeq: 61469 ACK. Max-Forwards: 70. Content-Length: 0. . # U xxx.xxx.190.248:5060 -> 61.229.4.212:5060 .... # U xxx.xxx.190.248:5060 -> 61.217.225.150:1720 .... # U xxx.xxx.190.248:5060 -> 61.229.4.212:35375 .... exit 30 received, 0 dropped

1 1

[Serusers] Confirming nathelper/rtpproxy versions
by Aisling O'Driscoll 25 Feb '05

25 Feb '05

Hi everyone, I just would like to clarify that the versions of nathelper and rtpproxy that I am running together are compatible. localhost:/ grep Id: rtpproxy/main.c * Id: main.c,v 1.21 2004/08/13 19:04:23 sobomax Exp$ localhost:/ head -1 sip_router/modules/nathelper/nathelper.c * Id: nathelper.c,v 1.77 2005/02/15 18:50:56 sobomax Exp$ Could someone confirm if the above work successfully together? I am running ser 0.8.14. Thanks, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

2 1

Re: [Serusers] SER SDP Port Problem??
by Greger V. Teigre 25 Feb '05

25 Feb '05

Yep. If you have configured nathelper to communicate with rtpproxy using udp, you can use tcpdump port 22222 (which is the default communication port, I believe) Also, rtpproxy can be started with -f option (no-fork) g-) Aisling O'Driscoll wrote: > Hi Greger, > > Thanks for the advice. If I run "netstat -tunap" should I see > rtpproxy with a PID and state listening?? How do I use tcpdump i.e. > what ports am I monitoring? > > RTPProxy is installed and I have provisions were made for it in my > original ser.cfg. However there still seems to be no success. I am > having my doubts about whether rtpproxy is actually functioning at > all. > > Thank you, > Aisling. > > > > ---- Original Message ---- > From: greger(a)teigre.com > To: ashling.odriscoll(a)cit.ie > Subject: Re: [Serusers] SER SDP Port Problem?? > Date: Mon, 21 Feb 2005 16:51:16 +0100 > >> If you have the latest cvs version, you can use "19" (add test=16). >> It may >> make a difference if you have configured your UA with stun. If not, >> "3" >> should be good. >> Use some sort of stun client implementation to test your nat. >> http://sourceforge.net/project/showfiles.php?group_id=47735&package_i >> d=102146 >> >> rtpproxy is rather silent. :-( You should do a tcpdump port >> control_port to >> see if any traffic is going. >> Also, you can do netstat -nlp to see if any udp ports are allocated >> once you >> believe a call is proxied. >> g-) >> >> Aisling O'Driscoll wrote: >>> Hi Greger, >>> >>> I only use nat_uac_test("3") as that is what is given in most >>> examples on the mailing list. Is there any way that i can test if I >>> am behind symmetric nat? I have rtpproxy running with ser but not >>> sure if my below ser.cfg script invokes it correctly. Do you know >>> where I could find the rtpproxy debug log? >>> >>> Thanks again, >>> Aisling. >>> >>> ---- Original Message ---- >>> From: greger(a)teigre.com >>> To: ashling.odriscoll(a)cit.ie, serusers(a)lists.iptel.org >>> Subject: Re: [Serusers] SER SDP Port Problem?? >>> Date: Mon, 21 Feb 2005 12:11:15 +0100 >>> >>>> Aisling, >>>> The port is in the m= line. You use nat_uac_test("3"). Any >> particular >>>> reason >>>> for not using all tests? >>>> Please note that sdp rewriting does not work if the UA is behind a >>>> symmetric >>>> NAT and you are calling in. You must then via an RTP proxy like >>>> mediaproxy >>>> or rtpproxy. >>>> g-) >>>> >>>> Aisling O'Driscoll wrote: >>>>> Hi, >>>>> >>>>> I understand the basic problem behind one way audio is that the >>>>> client behind nat has a private address in the sdp information, >>>>> therefore the voice cannot be delivered to this private address. >> It >>>>> is necessary to rewrite this sdp info with the nat public >> address. >>>>> However I have done this in my ser.cfg and I still have one way >>>>> voice. >>>>> >>>>> I have included some of relevant ethereal messages on SER and my >>>>> ser.cfg below. The messages show that the rtp information has >> been >>>>> changed. Does anyone think the problem is because there is no >> port >>>>> information in the "c" and "o" fields in the sdp?? If so how can >> I >>>>> make sure the port is included? >>>>> >>>>> Many Thanks, >>>>> Aisling. >>>>> >>>>> Ethereal messages: >>>>> >>>>> call between private client with public nat address 63.218.54.71 >>>> and >>>>> a public client with address 157.190.183.80. The SER address is >>>>> 157.190.183.70. >>>>> >>>>> REGISTER sip:157.190.183.70 SIP/2.0 >>>>> VIA: SIP/2.0/UDP 63.218.54.71:11987;rport;branch=z9h..... >>>>> FROM: whoever <sip:2008@157.190.183.70>;tag=455.... >>>>> TO: whoever <sip:2008@157.190.183.70> >>>>> CONTACT: "whoever" <sip:2008@63.218.54.71:11987> >>>>> Call-Id: .... >>>>> CSeq: 22227 REGISTER >>>>> Expires; 1800 >>>>> User Agent: X-Lite release 1103m >>>>> Content-Length: 0 >>>>> >>>>> SIP/2.0 200 OK >>>>> VIA: SIP/2.0/UDP 63.218.54.71:11987;rport;branch=z9h..... >>>>> FROM: whoever <sip:2008@157.190.183.70>;tag=455.... >>>>> TO: whoever <sip:2008@157.190.183.70> >>>>> CONTACT: "whoever" <sip:2008@63.218.54.71:11987>;q=0.00 >>>> expires=1800 >>>>> Call-Id: .... >>>>> CSeq: 22227 REGISTER >>>>> Expires; 1800 >>>>> User Agent: X-Lite release 1103m >>>>> Content-Length: 0 >>>>> >>>>> The public client (157.190.183.80 also registers) >>>>> >>>>> Then the private client invites the public client to a voice >>>>> conversation: >>>>> >>>>> INVITE sip:2001@157.190.183.70 SIP/2.0 >>>>> VIA: SIP/2.0/UDP 63.218.54.71:11987;rport;branch=z9h..... >>>>> FROM: whoever <sip:2008@157.190.183.70>;tag=455.... >>>>> TO: whoever <sip:2001@157.190.183.70> >>>>> CONTACT: "whoever" <sip:2008@63.218.54.71:11987> >>>>> Call-Id: .... >>>>> CSeq: 19929 INVITE >>>>> Expires; 1800 >>>>> User Agent: X-Lite release 1103m >>>>> Content-Type=application/sdp >>>>> Content-Length: 290 >>>>> >>>>> Session description Protocol >>>>> Owner/Creator of the Session (o): 2008 245812 272828 IN IP4 >>>>> 63.218.54.71 >>>>> Connection information (c): IN IP4 63.218.54.71 >>>>> >>>>> A 100 Trying is sent back from SER to the private client (i.e. >>>> caller) >>>>> >>>>> The INVITE is forwarded from SER to public client (callee) as >> show >>>>> below: >>>>> >>>>> INVITE sip:2001@157.190.183.70 SIP/2.0 >>>>> VIA: SIP/2.0/UDP 157.190.183.70;branch=.... >>>>> VIA: SIP/2.0/UDP 63.218.54.71:11987;branch=z9h..... >>>>> FROM: whoever <sip:2008@157.190.183.70>;tag=455.... >>>>> TO: whoever <sip:2001@157.190.183.70> >>>>> CONTACT: "whoever" <sip:2008@63.218.54.71:11987> >>>>> Call-Id: .... >>>>> CSeq: 19929 INVITE >>>>> Expires; 1800 >>>>> User Agent: X-Lite release 1103m >>>>> Content-Type=application/sdp >>>>> Content-Length: 290 >>>>> >>>>> Session description Protocol >>>>> Owner/Creator of the Session (o): 2008 245812 272828 IN IP4 >>>>> 63.218.54.71 >>>>> Connection information (c): IN IP4 63.218.54.71 >>>>> >>>>> 157.190.183.80 157.190.183.70 SIP 100 Trying >>>>> 157.190.183.80 157.190.183.70 SIP 180 Ringing >>>>> 157.190.183.70 63.218.54.71 SIP 180 Ringing >>>>> >>>>> 157.190.183.80 157.190.183.70 SIP/SDP Status: 200OK >>>>> >>>>> SIP/2.0 200 OK >>>>> Via: SIP/2.0/UDP 157.190.183.70;branch=.... >>>>> Via: SIP/2.0/UDP 63.218.54.71:11987;rport=11987;branch=..... >>>>> From: whoever<sip:2008@157.190.183.70>;tag=... >>>>> To: <sip:2001@157.190.183.70>;tag=.... >>>>> CSeq: 19929 INVITE >>>>> User Agent: Grandtsream BT100 1.0.5.18 >>>>> Contact: <sip:2001@157.190.183.80> >>>>> >>>>> Session description protocol >>>>> (c) IN IP4 157.190.183.80 >>>>> >>>>> >>>>> # >>>>> # $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" >>>>> >>>>> alias="157.190.183.70:5060" >>>>> >>>>> # ------------------ module loading >>>> ---------------------------------- >>>>> >>>>> # 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" >>>>> loadmodule "/usr/lib/ser/modules/nathelper.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 >>>>> #NB Had to up this value from 1 to 11 because reinvites were >>>>> bombarding called phone >>>>> modparam("rr", "enable_full_lr", 11) >>>>> >>>>> #!! Nathelper >>>>> modparam("registrar", "nat_flag", 6) >>>>> modparam("nathelper", "natping_interval", 30) #Ping interval 30 s >>>>> modparam("nathelper", "ping_nated_only", 1) #Ping only clients >>>>> behind NAT >>>>> >>>>> modparam("tm", "fr_inv_timer", 80) >>>>> >>>>> # ------------------------- 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; >>>>> }; >>>>> >>>>> #########################added for cit client behind nat >>>>> 09/02/05####################### >>>>> if (nat_uac_test("3")){ >>>>> if (method == "REGISTER" || ! search("^Record-Route:")){ >>>>> log("Log: Someone trying to register from private >> IP,rewriting\n"); >>>>> fix_nated_contact(); #Rewrite contact with source IP >>>>> 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(); >>>>> >>>>> # loose-route processing >>>>> if (loose_route()) { >>>>> #commented 11/02/05 >>>>> #t_relay(); >>>>> 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) { >>>>> >>>>> log(1,"into loop"); >>>>> if (method=="REGISTER") { >>>>> >>>>> # Uncomment this if you want to use digest authentication >>>>> # if (!www_authorize("157.190.183.70", "subscriber")) { >>>>> # www_challenge("157.190.183.70", "0"); >>>>> # break; >>>>> # }; >>>>> >>>>> save("location"); >>>>> break; >>>>> }; >>>>> >>>>> lookup("aliases"); >>>>> if (!uri==myself) { >>>>> append_hf("P-hint: outbound alias\r\n"); >>>>> route(1); >>>>> break; >>>>> }; >>>>> >>>>> if (method=="INVITE"){ >>>>> log(1,"in invite loop"); >>>>> #break; #no 100 trying >>>>> t_on_failure("1"); >>>>> }; >>>>> >>>>> # native SIP destinations are handled using our USRLOC DB >>>>> if (!lookup("location")) { >>>>> #sl_send_reply("404", "Not Found"); >>>>> route(2); >>>>> break; >>>>> }; >>>>> }; >>>>> >>>>> # forward to current uri now; use stateful forwarding; that >>>>> # works reliably even if we forward from TCP to UDP >>>>> #commented 11/02/05####################### >>>>> if (!t_relay()) { >>>>> sl_reply_error(); >>>>> }; >>>>> >>>>> } >>>>> >>>>> ######################################entered >>>>> >>>> >> 11/02/05############################################################ >>>>> 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; >>>>> }; >>>>> >>>>> t_on_reply("1"); >>>>> >>>>> if(!t_relay()){ >>>>> sl_reply_error(); >>>>> }; >>>>> } >>>>> ######################################entered >>>>> >>>> >> 11/02/05############################################################ >>>>> #!! Nathelper >>>>> onreply_route[1]{ >>>>> if(isflagset(6) && status =~ "(183)|2[0-9][0-9]"){ >>>>> fix_nated_contact(); >>>>> force_rtp_proxy(); >>>>> } else if (nat_uac_test("1")){ >>>>> fix_nated_contact(); >>>>> }; >>>>> } >>>>> >>>> >> ##################################################################### >>>> # >>>>> ########################################### >>>>> >>>>> # ------------- handling of unavailable user ------------------ >>>>> route[2] { >>>>> >>>>> # non-Voip -- just send "off-line" >>>>> if (!(method == "INVITE" || method == "ACK" || method == >>>>> "CANCEL")) { >>>>> sl_send_reply("404", "Not Found"); >>>>> break; >>>>> }; >>>>> >>>>> # forward to voicemail now >>>>> rewritehostport("157.190.183.70:5062"); >>>>> t_relay_to_udp("157.190.183.70", "5062"); >>>>> } >>>>> >>>>> # if forwarding downstream did not succeed, try voicemail running >>>>> # at 157.190.183.70:5062 >>>>> >>>>> failure_route[1] { >>>>> revert_uri(); >>>>> rewritehostport("157.190.183.70:5062"); >>>>> append_branch(); >>>>> t_relay_to_udp("157.190.183.70", "5062"); >>>>> } >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> -------------------Legal >>>>> Disclaimer--------------------------------------- >>>>> >>>>> The above electronic mail transmission is confidential and >> intended >>>>> only for the person to whom it is addressed. Its contents may be >>>>> protected by legal and/or professional privilege. Should it be >>>>> received by you in error please contact the sender at the above >>>>> quoted email address. Any unauthorised form of reproduction of >> this >>>>> message is strictly prohibited. The Institute does not guarantee >>>> the >>>>> security of any information electronically transmitted and is not >>>>> liable if the information contained in this communication is not >> a >>>>> proper and complete record of the message as transmitted by the >>>>> sender nor for any delay in its receipt. >>>>> >>>>> _______________________________________________ >>>>> Serusers mailing list >>>>> serusers(a)lists.iptel.org >>>>> http://lists.iptel.org/mailman/listinfo/serusers >>>> >>>> >>>> -------------------Legal >>>> Disclaimer--------------------------------------- >>>> >>>> The above electronic mail transmission is confidential and >> intended >>>> only for the person to whom it is addressed. Its contents may be >>>> protected by legal and/or professional privilege. Should it be >>>> received by you in error please contact the sender at the above >>>> quoted email address. Any unauthorised form of reproduction of >> this >>>> message is strictly prohibited. The Institute does not guarantee >> the >>>> security of any information electronically transmitted and is not >>>> liable if the information contained in this communication is not a >>>> proper and complete record of the message as transmitted by the >>>> sender nor for any delay in its receipt. >>>> >>> >>> >>> >>> -------------------Legal >>> Disclaimer--------------------------------------- >>> >>> The above electronic mail transmission is confidential and intended >>> only for the person to whom it is addressed. Its contents may be >>> protected by legal and/or professional privilege. Should it be >>> received by you in error please contact the sender at the above >>> quoted email address. Any unauthorised form of reproduction of this >>> message is strictly prohibited. The Institute does not guarantee >> the >>> security of any information electronically transmitted and is not >>> liable if the information contained in this communication is not a >>> proper and complete record of the message as transmitted by the >>> sender nor for any delay in its receipt. >> >> >> -------------------Legal >> Disclaimer--------------------------------------- >> >> The above electronic mail transmission is confidential and intended >> only for the person to whom it is addressed. Its contents may be >> protected by legal and/or professional privilege. Should it be >> received by you in error please contact the sender at the above >> quoted email address. Any unauthorised form of reproduction of this >> message is strictly prohibited. The Institute does not guarantee the >> security of any information electronically transmitted and is not >> liable if the information contained in this communication is not a >> proper and complete record of the message as transmitted by the >> sender nor for any delay in its receipt. >> > > > > -------------------Legal > Disclaimer--------------------------------------- > > The above electronic mail transmission is confidential and intended > only for the person to whom it is addressed. Its contents may be > protected by legal and/or professional privilege. Should it be > received by you in error please contact the sender at the above > quoted email address. Any unauthorised form of reproduction of this > message is strictly prohibited. The Institute does not guarantee the > security of any information electronically transmitted and is not > liable if the information contained in this communication is not a > proper and complete record of the message as transmitted by the > sender nor for any delay in its receipt.

1 0

Re: [Serusers] ser-0.8.14 / ser-0.9 + rtpproxy + PSTN
by support 25 Feb '05

25 Feb '05

----- Original Message ----- From: support To: Jefrey ; serusers(a)lists.iptel.org Sent: Friday, February 25, 2005 3:46 PM Subject: Re: [Serusers] ser-0.8.14 / ser-0.9 + rtpproxy + PSTN Hi, Yes, this configuration allow SIP clients to call each other. This is the updated config. Thomas # # $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 -----------

1 0

Re: [Serusers] ser-0.8.14 / ser-0.9 + rtpproxy + PSTN
by support 25 Feb '05

25 Feb '05

Hi, Yes, this configuration allow SIP clients to call each other. This is the updated config. Thomas # # $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 ----------- ----- Original Message ----- From: Jefrey To: 'support' Sent: Wednesday, February 23, 2005 10:44 PM Subject: RE: [Serusers] ser-0.8.14 / ser-0.9 + rtpproxy + PSTN Hi Just needing some of your expertise, does this configuration allow your SIP clients to call each other? Regards, Jef. ------------------------------------------------------------------------------ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of support Sent: Monday, February 14, 2005 5:42 PM To: serusers(a)lists.iptel.org Subject: [Serusers] ser-0.8.14 / ser-0.9 + rtpproxy + PSTN Hi, After studying some ser.cfg files, I could route a SIP call to PSTN gateway successfully. If ser receives "866+any PSTN num", ser will forward to T1 trunk gateway. T1 trunkgateway will drop the prefix 866 and route the call out to PSTN line. i.e. SIP client (866+PSTN num) ---> T1 Trunk Gateway (drop prefix 866) ---> PSTN call (PSTN num) SER version: ser-0.8.14 or ser-0.9. rtpproxy version: latest cvs version from berlios I have posted a ser.cfg in this email. Best Regards, Thomas # # $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 ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" # 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" # Nathelper loadmodule "/usr/local/lib/ser/modules/nathelper.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) # -- 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{ # 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; }; # 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:")) { log("LOG: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart enough to be symmetric. In some phones it takes a configuration # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is # called "symmetric media" and "symmetric signalling". fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("", "subscriber")) { www_challenge("", "0"); break; }; save("location"); break; }; if (uri=~"^sip:866*") { log(1, "going to PSTN route\n"); route(2); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # !! Nathelper if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); break; }; } route[2] { force_rtp_proxy(); record_route(); t_on_reply("1"); t_relay_to_udp("T1 gateway IP","T1 Gateway UDP port"); } # !! 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(); }; }

1 0

[Serusers] DNS SRV lookup issue
by Gandhi, Chetan 25 Feb '05

25 Feb '05

Hi All, I am facing problem to get SER to do a SRV lookup. I have the following scenario... Step 1: SIP UA1 registers with SER (testdomain.com) with contact address user(a)testdomain.com SIP UA2 registers with SER (testdomain.com) with contact address user(a)testdomain.com Step 2: SIP UA3 calls user(a)testdomain.com Problem I would expect SER to do a DNS SRV lookup on testdomain.com and get sip1.testdomain.com and sip2.testdomain.com. Unfortunately, SER only does A and AAAA query. (verified via ethereal) and does not do SRV lookup. How can I make it do SRV lookup ? I am using the default ser.cfg file. DNS entries _sip._udp.testdomain.com. 3600 IN SRV 0 1 5060 sip1.testdomain.com. _sip._udp.testdomain.com. 3600 IN SRV 0 1 5060 sip2.testdomain.com. I am trying to do load balancing but the SRV lookup failure has got me stuck. Any pointers would be really appreciated. Thanks, Chetan

3 3

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users