sr-users June 2004

sr-users@lists.kamailio.org

176 participants
387 discussions

by Zappasodi Daniele

Suppose this simply scenario: 3 SIP phones A,B,C. On B is set a redirection (moved temporarily) direct to C. On C is set a redirection (moved temporarily) direct to B. <?xml:namespace prefix = o ns = "urn:schemas-microsoft-com:office:office" /> If A calls B and B replies with moved temporarily to C. A INVITE B A ---------a B B “Moved temporarily to C” A s--------- B A calls C and B replies with moved temporarily to B. A INVITE C A ---------a C C “Moved temporarily to B” A s--------- C A calls B A INVITE B A ---------a B so it generates an endless loop. How can I detect and avoid this endless loops? Does the INVITE contain an header field, analogous to the “Max Forwards”, that I can see to identify a loop and consequently refuse the INVITE? Thanks for your help, Daniele

20 years

[Serusers] PABX to SIP Server

by Ronald Ramos

Hi, Has anyone able to integrate a PABX to talk to a SIP server? So that I can use their existing phone system, but when they dial outside their network it will be the SIP server handling it. Thank you Ronald

20 years

[Serusers] Cisco Unity 4.0.4 software + SER proxy ( rtpproxy + nathelper + radius ) and it is running fine.

by ito

http://www.alpha-broadbusiness.jp/ We have the Cisco Unity software Cisco Unity 4.0.4 software + SER proxy ( rtpproxy + nathelper + radius ) and it is running fine.

20 years

Re: [Serusers] Click To Dial dialog mismatch

by wangji

Hi, I am testing click to talk function working with SER server like below: SER server -------------------------------------- | | Web server(click to talk by using click_to_dial function) | | make call | Microsoft IE (working as caller) ----------------> IAD or other soft client (working as callee) I wonder click to dial function in Serweb can work independency and how to deal with media stream(voice data). Jimmy ----- Original Message ----- From: "Jiri Kuthan" <jiri(a)iptel.org> To: "Ezequiel Colombo" <ecolombo(a)arcotel.net>; <serusers(a)iptel.org> Sent: Saturday, May 15, 2004 5:53 AM Subject: Re: [Serusers] Click To Dial dialog mismatch > At 10:13 PM 5/14/2004, Ezequiel Colombo wrote: > >Hi, i am trying the funcion click_to_dial included in the serweb package and i note that the function stored in the functions.php file as "click_to_dial()" is not coded to send the ACK after the receipt of 200 OK from the client. My client (X-Lite) remain sending the 200OK forever and dont process the REFER message sent by the server. > > > >In the 72 page of the SER Admin Guide i see the dialog of click to dial function like this : > > > > Example 4-12. Call-Flow for Click-To-Dial Using REFER > > CTD Caller Callee > > #1 INVITE > > -----------------> > > ... > > caller answers > > #2 200 > > <----------------- > > #3 ACK > > -----------------> > > #4 REFER > > -----------------> > > #5 202 > > <----------------- > > #6 BYE > > -----------------> > > #7 200 > > <----------------- > > #8 INVITE > > ------------------> > > #9 180 ringing > > <------------------ > >Are the function included in SERWEB working fine on your systems ??? > > Yes. > > ACK is generated autonomously be SER's TM module. > > Perhaps SER logs will help you further. > > -jiri > > > > >Thanks > >Ezequiel Colombo > >_______________________________________________ > >Serusers mailing list > >Serusers(a)iptel.org > >http://mail.iptel.org/mailman/listinfo/serusers > > -- > Jiri Kuthan http://iptel.org/~jiri/ > > _______________________________________________ > Serusers mailing list > Serusers(a)iptel.org > http://mail.iptel.org/mailman/listinfo/serusers

20 years

[Serusers] Need help or i give up

by Sanjay Duggal

Hi All I have been working to make this work. I have an Asterisk gateway and a Ser proxy running on to different servers. Ser has one Public Ip and a private one. ( on two different NICs) I have installed RTPProxy on the same server as Ser. When I make a call from my cell phone I can have a conversion on my SIP phone. Everything works great. But when I call from SIP phone it is no sound. My cell rings I can pick it up but no sound. Can one of you expert on this mater please help me? I'm enclosing my ser.cfg # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=4 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=9 fork=yes log_stderror=yes */ alias="Domain.com" alias="my.Domain.com" # alias="192.168.0.100" # alias="192.168.0.200" listen="public ip" 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" fifo_mode=0777 # ------------------ 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" loadmodule "/usr/local/lib/ser/modules/nathelper.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) # !! 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 # ------------------------- 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 # 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 is used); also, # the received test should, if completed, should check all # vias for rpesence of received if (nat_uac_test("3")) { log(1, "NAT client\n"); record_route(); # 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") { if (method == "REGISTER" || !search("^Record-Route:")) { log(1, "LOG: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that supportsymmetric # communication. We tested quite many of them andmajority is # smart enough to be symmetric. In some phones it takesa 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 IPof signalling if (method == "INVITE") { log(1, "NAT -> INVITE\n"); fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed append_to_reply("P-NATed-Caller: Yes\r\n"); }; }; # 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=~"^sip:0[0-9]*@my.Domain.com") { log(1, "Forwarding to Asteriks\n"); route(1); # rewritehostport("192.168.0.200:5060"); # append_hf("P-hint: GATEWAY\r\n"); # t_relay_to_udp("192.168.0.200", "5060"); #forward(192.168.0.200,5060); # Where local asterisk is listening #t_relay(); 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") { log(1, "Myself -> REGISTER\n"); # Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "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; }; }; #inserted by klaus if (method=="INVITE") { log(1, "INVITE\n"); record_route(); force_rtp_proxy(); /* set up reply processing */ t_on_reply("1"); }; # 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(); }; # append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # !! Nathelper log(1, "ROUTE[1]\n"); 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 IPaddresses"); break; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { log(1, "Flag is 6 (NAT)\n"); if (!is_present_hf("P-RTP-Proxy")) { force_rtp_proxy(); append_hf("P-RTP-Proxy: YES\r\n"); }; append_hf("P-NATed-Calee: Yes\r\n"); rewritehostport("192.168.0.200:5060"); append_hf("P-hint: GATEWAY\r\n"); t_relay_to_udp("192.168.0.200", "5060"); break ; }; # NAT processing of replies; apply to all transactions (forexample, # 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()) { log(1, "ROUTE[1] -> sl_reply_error\n"); sl_reply_error(); break; }; } # !! Nathelper onreply_route[1] { log(1, "onreply_route[1]\n"); if (status=~"[12][0-9][0-9]"){ # force_rtp_proxy(); # if (isflagset(6) && status=~"(183)|2[0-9][0-9]") { fix_nated_contact(); fix_nated_sdp("1"); force_rtp_proxy(); } else #if(nat_uac_test("1")) { fix_nated_contact(); force_rtp_proxy(); }; } regards ------------------------------------------------------------- Pressis Consulting DA Sanjay Duggal Operative CTO --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 15.06.2004

20 years

[Serusers] failure route and ERROR: t_should_relay: statusrewrite by UAS: stored: 408, received: 487

by Steve Blair

Hello Jiri: I saw the following post with your name assocated with it.I did not see a resolution to the initial question, that is why is this message being generated? [Serusers] failure route and ERROR: t_should_relay: status rewrite by UAS: stored: 408, received: 487 <http://lists.iptel.org/pipermail/serusers/2003-December/004635.html> In my case the call is being forwarded to an Octel voice mail system that is reachable by going through a Cisco router. Can you add some insight into what this message means and why Jun 17 15:18:15 ser[21652]: ERROR: t_should_relay: status rewrite by UAS: stored: 408, received: 487 The sections of my config file being touched follow: # redirect user to vm if not availabile? Flag it now in case of # rewrite and store it in flag 6 for route block 6 if (is_user_in("Request-URI", "voicemail")) { t_on_failure("6"); setflag(6); log(1, "[SER]: Flag for VM redirect successful. \n"); } else { t_on_failure("1"); log(1, "[SER]: Flag for VM redirect unsuccessful. \n"); }; # failure_route[6] { xlog("L_INFO", "\n[SER]: Failure Route block #6 Unavailable user: Time: [%Tf] Method: <%rm> R-uri: <%ru> Contact Header: <%ct> From uri <%fu> To < %tu> IP source address <%is> \n\n"); append_branch("sip:86423@130.1.5.3"); append_urihf("CC-Diversion: ", "\r\n"); append_hf("P-hint: OFFLINE-VOICEMAIL\r\n"); t_relay(); }

20 years

[Serusers] RADIUS support

by AJ Grinnell

Getting the error radclient:No token read where we expected an attribute name when testing the RADIUS install. Saw in past archieves that it could be a problem with # and comments, so I removed all of them, any other ideas? AJ Grinnell

20 years

[Serusers] Error on start ser.

by gc

After starting ser, I run ser status and got following message: 'ser dead but subsys locked' How can I fix this? I remember in the e-mail archive somewhere metioned that you can clean some files to correct this. I can not find it now. Gary

20 years

Re: [Serusers] DB Error(table admin_privileges not exist) when usingserweb

by wangji

I find the reason. The version of SER that I used is branch 0.8.12 from CVS server. But SERWEB is compliant with CVS branch 0.8.13. The file ser_mysql.sh from branch 0.8.13 create table admin_privileges. So are there a branch of SERWEB support SER 0.8.12 ? Jimmy ----- Original Message ----- From: "wangji" <wangji(a)bjut.edu.cn> To: <serusers(a)iptel.org> Sent: Thursday, June 17, 2004 7:39 PM Subject: [Serusers] DB Error(table admin_privileges not exist) when usingserweb > Hi all, > I got a DB error when I tried to login into serweb(both admin and user_interface). > The message showing on IE is "DB Error: no such table, file: /var/www/novsky.com/serweb/html/data_layer.php:405". > I checked mysql's log. It query DB ser, is: > select priv_name, priv_value from admin_privileges where username = 'admin' and domain = 'iptel.org' and (priv_name = 'change_privileges' or priv_name = 'is_admin'); > That table "admin_privileges" is not exist. > In DB ser, there are these tables: > +-----------------------+ > | Tables_in_ser | > +-----------------------+ > | acc | > | active_sessions | > | aliases | > | config | > | domain | > | event | > | grp | > | location | > | missed_calls | > | pending | > | phonebook | > | preferences | > | reserved | > | server_monitoring | > | server_monitoring_agg | > | silo | > | subscriber | > | uri | > | version | > +-----------------------+ > And these tables is created by ser_mysql.sh. > How can I do? > > > Best regards, > > Jimmy > > -------------------------------------------------------------------------------- > _______________________________________________ > Serusers mailing list > Serusers(a)iptel.org > http://mail.iptel.org/mailman/listinfo/serusers >

20 years

[Serusers] Creating a conference with SEMS

by langals＠mweb.co.za

20 years

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sr-users June 2004