sr-users September 2005

sr-users@lists.kamailio.org

231 participants
507 discussions

[Serusers] 483 Too Many Hops
by BOUNASREDDINE Samer 05 Sep '05

05 Sep '05

Hi, I'm working on a project and i'm using ser as a sip server and webser as it's web interface. actually i have this problem: every time i send an instant message to a subscriber i get this error message: 483 too many hops i tried to fix it but i couldn't and i don't know what to do ! if any one can help me i'll be very grateful thank you in advance samer i'm using ser 0.9.3 & here's my code: # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 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 #fork=no log_stderror=yes check_via=no # (cmd. line: -v) dns=yes # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" fifo_mode=0666 fifo_db_url="mysql://ser:password@localhost/ser" alias=domain.com # ------------------ 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/pa.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_url", "mysql://ser:password@localhost/ser") #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", "secret", "123123123") modparam("auth_db", "db_url", "mysql://ser:heslo@localhost/ser") 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("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 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("domain.com", "subscriber")) { www_challenge("domain.com", "0"); break; }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; }

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[Serusers] Route forward on condition
by peterb 05 Sep '05

05 Sep '05

I am not sure where I am failing on the expressions. What I am trying to do is if the source Ip is correct along with the methods I want to send it to a possible of two places one is to a pstn if it is not for the pstn maybe they are trying to reach a user that is registered with the local domain. It works on the pstn. On the local domain even if the src_ip is wrong it still forwards the request as long as it sees the prefix 555555. I want the src_ip to be correct also on the request. -------- snippet -------- if (src_ip==xxx.xxx.xxx.xxx & method=="INVITE" || method=="ACK" || method=="BYE") { if (uri=~"^sip:55555[0-9]{11}@") { strip(5); setflag(1); record_route(); rewritehost("xxx.xxx.xxx.xxx"); #pstn }; if ( (uri=~"^sip:55555[0-9]{7}@") | (uri=~"^sip:55555[0-9]{10}@") ) { strip(5); record_route(); route(2); # local users break; }; } ------------ snippet end ---------------------- Would also like to be able to relay FWD requests Any thoughts ? Thanks Mr. B

1 0

[Serusers] siproxy howto
by crazy 05 Sep '05

05 Sep '05

Hi all What I can do for ser, let it have the function of proxy , let it can pass throught the firewall or NAT

1 0

[Serusers] SER + Mediaproxy /RTProxy
by GTel 05 Sep '05

05 Sep '05

Hello all, I plan to use SER with MediaProxy as a far-end NAT traversal solution. As much as I understand, SER Proxy server does the actual RTP traffic forwarding between the parties. Using SER with MediaProxy only as a far-end NAT traversal solution (or using RTProxy) can I forward register to a 3rd party Proxy /registrar or the nated parties have to register with SER ? Thanks for any help, GTel

1 0

FW: [Users] t_relay does not follow RFC3263 "Locating SIP Servers"
by Franz Edler 04 Sep '05

04 Sep '05

An additional remark: What is the difference between t_relay and t_relay_to_udp when t_relay does not use NAPTR and does only an SRV lookup towards udp? regards Franz _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

2 1

[Serusers] MWI - message waiting indication
by harry gaillac 03 Sep '05

03 Sep '05

hello, I read http://www.voip-info.org/tiki-index.php?page=Asterisk+at+large anybody could tell me more about this ? Is it available with ARA ? Regards Harry Method 3 Q: If you have your SIP phones registered with SER but your voicemail is handled by asterisk, how do you get the MWI (Message Waiting Indicator) light to function on the phone? A: In sip.conf create a section pointing at your SER router. [ser] type=friend ; We allow incoming and outgoing calls. Use peer if you are only doing MWI context=ser ; This is the context incoming calls land in host=ser.server.tld ; This is the hostname or IP address of your SER server fromdomain=ser.server.rld ; This is your SER_DOMAIN insecure=very ; This allows incoming calls from the phones routing through ser to be passed into asterisk mailbox=user@context ; This is where you list the voicemail boxes to monitor This tells asterisk that if a voicemail comes in to "user" then it needs to send a SIP NOTIFY message to the "ser.server.tld" phone. Well this is all well and good except how does SER deliver this NOTIFY to the phones? First thing is that you need to make a tiny change to the asterisk code to pass the mailbox user in the SIP NOTIFY packet. --- channels/chan_sip.c.orig Thu Jul 14 12:03:18 2005 +++ channels/chan_sip.c Thu Jul 14 12:05:26 2005 @@ -9710,6 +9710,7 @@ /* Called with peerl lock, but releases it */ struct sip_pvt *p; int newmsgs, oldmsgs; + char *s; /* Check for messages */ ast_app_messagecount(peer->mailbox, &newmsgs, &oldmsgs); @@ -9735,6 +9736,10 @@ /* Recalculate our side, and recalculate Call ID */ if (ast_sip_ouraddrfor(&p->sa.sin_addr,&p->ourip)) memcpy(&p->ourip, &__ourip, sizeof(p->ourip)); + strcpy(p -> username, peer -> mailbox); /* Username = Mailbox name */ + s = strchr(p -> username, '@'); /* Remove the context part */ + if (s != NULL) + *s = 0; build_via(p, p->via, sizeof(p->via)); build_callid(p->callid, sizeof(p->callid), p->ourip, p->fromdomain); /* Send MWI */ After this patch is applied, the MWI NOTIFY messages coming from asterisk will have the URI user(a)ser.server.tld. This can be then routed with ser to the correct phone with normal SER routing rules. ie. SER does a lookup("location") and then a t_relay(). I don't believe this patch should effect any non-ser controlled sip phones. For me, this method was a lot easier then Method 2 listed above. You can add as may mailbox's as you like into the mailbox= line in the asterisk sip.conf file. One possible problem is if you have a mailbox called 1000@cx1 and another called 1000@cx2, this patch will make the MWI indicator light up for phone 1000(a)ser.server.tld when either mailbox gets a message. A simple modification to the patch and SER could be used to handle multiple contexts if required however this simplification is sufficient for me. ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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[Serusers] Compile problems on Mac OS X Tiger
by Hans Eriksson 03 Sep '05

03 Sep '05

Hi, I am trying to compile ser on my Mac OS X but it fails with: ... gcc -bundle -bundle_loader ../../ser db_con.o db_mod.o dbase.o my_con.o my_id.o my_pool.o res.o row.o utils.o val.o -L/usr/lib/ mysql -L/usr/local/lib -L/usr/local/lib/mysql -L/usr/local/mysql/lib/ mysql/ -L/usr/local/mysql/lib -L/usr/lib64/mysql -lmysqlclient -lz -o mysql.so ld: warning -L: directory name (/usr/lib/mysql) does not exist ld: warning -L: directory name (/usr/local/mysql/lib/mysql/) does not exist ld: warning -L: directory name (/usr/lib64/mysql) does not exist ld: Undefined symbols: _fprintf$LDBLStub _sprintf$LDBLStub _printf$LDBLStub _vsprintf$LDBLStub make[1]: *** [mysql.so] Error 1 ... All the other modules compile and link fine. Anyone got any ideas? /hans [kuben:~/ser2] hans% gcc -v Reading specs from /usr/libexec/gcc/darwin/ppc/3.3/specs Thread model: posix gcc version 3.3 20030304 (Apple Computer, Inc. build 1495) ... Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 450 to server version: 4.1.14-standard

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[Serusers] jabber module
by harry gaillac 03 Sep '05

03 Sep '05

Hello, Did anybody try Jive Messenger server http://www.jabber.org/software/JiveMessenger.shtml with ser ? Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] SER+ASTERISK voicemail
by harry gaillac 02 Sep '05

02 Sep '05

Hello, I set SER as sip proxy and ASTERISK as voicemail server (ARA) and serweb as TUI (telephone user interface) . Serweb | Ua-------ser-------asterisk voicemail | | Mysql DB I add user agents with address sip:name@domain + aliases sip:123@domain where 123 is mailbox I can forward voice messages to Asterisk with "failure route" for status 408 or 486. However I can't do it for offline users because of SER look for addresses like sip:user@domain not sip:123@domain where 123 is mailbox How could I solve this problem if possible ? Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Users] BYE before 200 OK?
by Michael Ulitskiy 02 Sep '05

02 Sep '05

Hello, Can someone please enlighten me on if this is allowed to send BYE before 200 OK? I thought that BYE should be used only after dialog established, but now I have a SIP PSTN termination provider and they appear to be sending BYE with route headers right after 183 Session Progress in some cases. So the session looks like the following: UAC OpenSER PSTN Provider UAS INVITE to openser 180 Trying back to UAC INVITE to UAS 180 Trying back to openser 183 Session progress back to openser 183 Session Progress back to UAC BYE to openser with Route headers BYE to UAC (loose_routing) 200 OK to openser 200 OK to UAS Is it ok? And there's even more to it. I don't know if above is OK or not, but I would thought that now the session is ended. Nonetheless in exactly 50 seconds I'm getting 183 Session progress back to openser 183 Session progress back to UAC -- in another 60 seconds 183 Session progress back to openser 183 Session progress back to UAC -- this happens 6 times every 60 seconds and then 408 Request Timeout back to openser ACK to UAS 408 Request Timeout to UAC ACK to openser Do you think it's a provider bug? Another question is how openser is routing replies that cannot be matched to transaction as I think transaction has already been deleted after BYE and latest 183 replies were routed by some other principle. Could please someone comment on this? Thanks, See you later, Michael

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sr-users September 2005