sr-users January 2004

sr-users@lists.kamailio.org

142 participants
338 discussions

[Serusers] more about record route
by Juha Heinanen 11 Jan '04

11 Jan '04

i based my reasoning regarding the need to add record route only to the initial request to this text in rfc3261. i guess the proxy may add record route to any request, but since it doesn't have any effect to the dialog state, i don't see a point of doing it. -- juha 12.2 Requests within a Dialog Requests within a dialog MAY contain Record-Route and Contact header fields. However, these requests do not cause the dialog s route set to be modified, although they may modify the remote target URI. Specifi- cally, requests that are not target refresh requests do not modify the dialog s remote target URI, and requests that are target refresh requests do. For dialogs that have been established with an INVITE, the only target refresh request defined is re-INVITE (see Section 14). Other extensions may define different target refresh requests for dialogs established in other ways. Note that an ACK is NOT a target refresh request. Target refresh requests only update the dialog s remote target URI, and not the route set formed from the Record-Route. Updating the latter would introduce severe backwards compatibility problems with RFC 2543-compliant systems.

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[Serusers] RTPProxy compilation
by Adrian Georgescu 11 Jan '04

11 Jan '04

Use the alternative SER Media proxy which runs without recompilation (SER stable version only) http://SERMediaProxy.dns-hosting.info/ --- Adrian Georgescu www.ag-projects.com Message: 11 Date: Sat, 10 Jan 2004 21:48:11 -0300 From: rfrez(a)vtr.net Subject: [Serusers] Problems compiling main.c 1.2 RTPPROXY To: serusers(a)lists.iptel.org Message-ID: <3FE9489100021E5B(a)hudson.vtr.net> Content-Type: text/plain; charset="ISO-8859-15" Im unable to compile, because send a warning about a : "sa_len is not defined in structure (socketaddr)" Im using Suse 9, maybe Im missing some libs, or need an update???

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[Serusers] Headline news - NEW Alternative To Collagen Lip Injections
by Yppe Wv 11 Jan '04

11 Jan '04

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[Serusers] Problems compiling main.c 1.2 RTPPROXY
by rfrez＠vtr.net 11 Jan '04

11 Jan '04

Im unable to compile, because send a warning about a : "sa_len is not defined in structure (socketaddr)" Im using Suse 9, maybe Im missing some libs, or need an update???

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[Serusers] STUN Server Problem
by Shirley Toh 11 Jan '04

11 Jan '04

Hi, I have problem installing and run the STUN server. Error as below:~ # service stund start Starting /usr/sbin/stund: Port 10000 for receiving UDP is in use Port 10001 for receiving UDP is in use stund: stun.cxx:365: void stunServer(StunAddress, StunAddress, bool): Assertion `myFd != INVALID_SOCKET' failed. [FAILED] # /usr/sbin/stunq sipserver.com:5060 Internet connection is type: Blocked Please advice why the service when start have problem. and why the internet connection is blocked? Please help and any ways to encounter the problem of symmetical NAT and Full cone NAT, when my user is behind the Symmetrical firewall there is one way audio but for full cone NAT is okie. any solution to this problem? Thank you in advance.... regards, shirley

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[Serusers] Call Forward Uncontitionally
by Marcello Lupo 10 Jan '04

10 Jan '04

Hi to all, i have to implement the call transfer service for our offices using SER. We have a PSTN gateway (cisco 3725) that receive calls from PSTN to out VOIP equipments. I have already implemented an interface to let the user to activate the CFU and set it in a database. I made an external script to check this database (made in php). If is the CFU it active it rewrite the uri with the new destination number. Now i have accomplished to all work but i don't know how to make an IF statement based on a result of an esternal script i.e.: if(external_script==1) do the rewriting; else continue with the traversal of the rules. I need to do this in this way becouse after this check i have other rules that manage outgoing calls from VOIP so i need that if the CFU is not active the next rule have to be checked to determine the real destination address. After this i have another problem: If the call transfer is active the source number is leaved as is and the central Phone Switch do not allow to pass that number as source number, so it keep the default number (or anonymous) to originate the call. This becouse being on the client side we cannot generate the calls with all the source we want. I place the example: A) Calling number: 123456 (from PSTN) B) Called Number: 555555 (on VOIP) C) Transferred number : 999999 (on PSTN selected by VOIP user 555555 to transfer the call to) D) Default Number of the PRI : 111111 I am the user B and activate the call transfer to C. A call come from A to B. The SIP proxy receive the call and change the destination from B to C and mantain the source as A. When the Cisco generate the call to C it keep as source A and destination C. The problem is that A is not allowed as outgoing number from the PRI in this way the switch take the default number D as source and the customer on C see the D number as originator. This is not a wanted behaviour becouse the user is not able to know he is calling him. I know that there is a way to change the answer of SIP to insert B as source and C as destination and A as CC-Diversion in the sip request. In this way the cisco will put in the ISDN setup the redirecting number A and the switch box will interpret it as source. I have not so much experience in SIP Proxy to do it. I hope someone can help me. thanks, regards particularly to Ulrich that helped me a lot in the past. Bye, MArcello

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[Serusers] SIP to H323 Proxy/Gateway?
by Arnd Vehling 10 Jan '04

10 Jan '04

Hello, is there a piece of software which will connect a SIP Proxy to a H323 Gatekeeper/Softswitch? We want to route SIP calls into a H323 network for PSTN termination and and call routing to H323 endpoints within the H323 network. We are currently testing Vovidas siph323csgw but with no avail. Has anyone a similiar setup? If yes, which software youre using? best regards, Arnd

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[Serusers] Reg. SIP->H323 Converter
by Balaji Bapulal Thoguluva 10 Jan '04

10 Jan '04

Hi, Has anyone tried SER with SIPH323 protocol translator from sipquest.com? for making SIP->H323 calls and viceversa. If anybody has tried, could you tell how to configure SER to work properly with SIPH323?. I get error messages in SER while I call from SIP->Cisco skinny phones. Thanks, balaji

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[Serusers] REGISTER implementation --
by Kannaiyan Natesan 09 Jan '04

09 Jan '04

I was going through the source code of SER to implement REGISTER METHOD. Is there anyone already implementing this functionality? Kannaiyan

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[Serusers] Cant make registration , maybe conf problem
by rfrez＠vtr.net 09 Jan '04

09 Jan '04

I cant make registration with a sip client... my server doesnt throw errors... and is running, but I cant make a registration... here is my ser.cfg: I have one user : serctl add rfrez rfrezpwd rfrez(a)email.net and I tested with windows messenger with rfrez(a)172.16.6.84 as account, and 172.16.6.84 as server with udp ------ # # $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 ---------------------------------- # 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" # 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) modparam("usrloc", "db_url","sql://ser:heslo@localhost/ser") # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "db_url","sql://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 alias="172.16.6.84" 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=~"172.16.6.84") { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("172.16.6.84", "subscriber")) { www_challenge("172.16.6.84", "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(); }; }

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sr-users January 2004