sr-users April 2006

sr-users@lists.kamailio.org

188 participants
431 discussions

R: [Users] using AVPs to write into database

by D'Addelfio Davide

Hi guys, I have a similar problem, maybe easier. I'm new in openser so I'd like to have some help. I'd like openser create a table in MySQL db when receive an INVITE, a row with from uri, method and something else, I've look into modules docs but I've not understanding at all...could you help me, please? Thanks for reply Davide ________________________________ Da: users-bounces(a)openser.org [mailto:users-bounces@openser.org] Per conto di Jayesh Nambiar Inviato: mercoledì 5 aprile 2006 15.53 A: Bogdan-Andrei Iancu Cc: openser Oggetto: Re: [Users] using AVPs to write into database Hi Bogdan, Thanks for the reply and sorry for sending the earlier mail 4 times as yahoo was not responding peoperly. When I do an avp_write(), why does the avp name get stored in the uuid column? I am not using any uuid as of now. I also tried to add the username of the person who is calling but it again creates two rows, one with the value as username and other row with the value as forwarded number. I thought if there is a way to insert the username also, it wud be easier to check, if the incoming call for that user is to be forwarded or not. I get the following in my usr_preferences table after I dial 86 followed by any number. +-----------+----------+--------+-----------+------+-----------------------------+---------------------+ | uuid | username | domain | attribute | type | value | modified | +-----------+----------+--------+-----------+------+-----------------------------+---------------------+ | s:callfwd | | | callfwd | 0 | sip:5515551478@202.80.61.10 | 2006-04-05 19:10:49 | +-----------+----------+--------+-----------+------+-----------------------------+---------------------+ The script snippet is as follows: if(uri=~"^sip:86[0-9]*@") { strip(2); avp_write("$ruri", "s:callfwd"); #avp_write("$from/username", "i:999"); #avp_db_store("i:999", "i:/usr_preferences"); avp_db_store("s:callfwd", "s:callfwd/usr_preferences"); sl_send_reply("200", "OK"); exit; }; Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: Hi, you need two steps: 1) write the ruri (after strip) into an AVP - use avp_write() 2) write the avp into the db - use avp_db_store() regards, bogdan Jayesh Nambiar wrote: > Hi all, > I am trying to implement the call forwarding feature in openser. The > forwarding part is fine. But I have the follwing scenario: > Whenever a user needs to change his forwarding number, he should be > able to do so from his end device only. > For eg: He first presses 86 and then the 10 digit number to be > forwarded. Here the 86 has to be stripped and the 10 digit number > should be inserted into the call forward value column of the > preferences table. > Also if he wants to remove the forwarded number, he can do so by > pressing say for eg: 87. on receiving this number openser should > delete that value from the > table. > Is this possible. I tried to do the following, but somehow it does not > change the > value. > if(uri=~"^sip:86[0-9]*@") { > if(avp_db_load("$from/username", "s:callfwd")) { #check > if call-fwd feature is enabled for the user > log(1,"AVP condition returned true"); > strip(2); > avp_write("$ruri", "s:callfwd"); > avp_print(); > log(1,"AVP written"); > sl_send_reply("200", "OK"); > exit; > }; > }; > Is avp_write the proper method or I guess avp_db_store can also help > me. The avp_print() function also does not show me anything in the > log. Are there any > logical mistakes or I have mis-interpreted the syntax of avpops functions. > Please help me in thsi regard. > Thanks a lot in advance. > Jayesh. > > ------------------------------------------------------------------------ > Jiyo cricket on Yahoo! India cricket > > Yahoo! Messenger Mobile > > Stay in touch with your buddies all the time. > >------------------------------------------------------------------------ > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > ________________________________ Jiyo cricket on Yahoo! India cricket <http://us.rd.yahoo.com/mail/in/mailcricket/*http:/in.sports.yahoo.com/crick…> Yahoo! Messenger Mobile <http://us.rd.yahoo.com/mail/in/mailmobilemessenger/*http:/in.mobile.yahoo.c…> Stay in touch with your buddies all the time.

18 years, 1 month

[Serusers] Changing the destination port

by Bruno Machado

Hi all We are trying to install a firewall in our structure. The problem is: the firewall (between X.Y.Z.A and X.Y.Z.B) manipulates the packets, but it changes de SRC port when it sends to the SER (X.Y.Z.B). X.Y.Z.A:5060 ----> X.Y.Z.B:5060 (dump in X.Y.Z.A) X.Y.Z.A:10080 ----> X.Y.Z.B:5060 (doing a dump in X.Y.Z.B) When X.Y.Z.B answer to X.Y.Z.A, it uses the port 10080: X.Y.Z.B:5060 ----> X.Y.Z.A:10080 It break the comunication, because X.Y.Z.A isnt listening at 10080. I need to force the SER to send the messages back to 5060. There are any way to do it? Thanks a lot. Bruno Machado __________________________________________________ Faça ligações para outros computadores com o novo Yahoo! Messenger http://br.beta.messenger.yahoo.com/

18 years, 1 month

[Users] Use STUN always - is there any downside?

by Barry Flanagan

Hi, We are using SIPPS as the softphone for our ISP customers, all of whom are on DSL and behind some sort of NAT'ed firewall. SIPPS does not always pick up the need for STUN when it checks the network, and then the user has problems until STUN is enabled. My question is - would there be any downside in us just hard coding the client to use STUN, given that the end user is a home user on DSL? Thanks. -- -Barry Flanagan

18 years, 1 month

[Serusers] Managing callerid for incoming PSTN

by dashy dude

Dear All I am new to SER and wanted to figure out how to manage callerid when displayed on a end phone. I have a E1 line terminating on a Cisco Router which we use for inbound/outbound calls. We dial 9 to make a outbound call via this e1. When a incoming call comes, it displays the callerd as presented by my operator. Now, if the user wants to call back, he cant do by the menu of callback since dialing out from pstn needs a digit 9 prefix. I would like to know if I can prefix 9 before forwarding call to PSTN so that the callback from the extension works smoothly. Thanks in advance. Vish __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

18 years, 1 month

[Users] Best way to replace digits in dst_uri

by Barry Flanagan

Hi, I need to rewrite the destination uri to replace "+353" with "0" before passing it on to our PSTN gateway.. What is the cleanest way to do this? Thanks. -- -Barry Flanagan

18 years, 1 month

[Serusers] Config. SER on SuSE Linux

by Jahangir Mushtaq

My name is Jahangir I am configuring SER on SuSE Linux.I have some problem to configure it.Any one wants to help me? Jahangir __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

18 years, 1 month

[Users] syn_branch

by Simon Morvan

Just to have the confirmation : When acting as a stateless proxy (not using tm at all), the variable syn_branch have to be 0. syn_branch = 1 is meaningful only in case we use tm ? And the documentation says the default value of syn_branch is 0 but the code initialize the variable to 1. I have to manually set it to 0. -- Simon.

18 years, 1 month

[Users] ERROR: t_newtran: transaction already in process - using UAC module

by Cameron Beattie

I am trying to use the UAC module in conjunction with a gateway that requires authentication. I am getting the following when trying to dial a number that gets sent to the gateway: ERROR: t_newtran: transaction already in process Prior to this the following are written to the log file: international PSTN match resetflag(9) host rewritten What seems to be happening is that the t_relay_to_udp is being called and the gateway is contacted but before any reply is received openser is throwing a t_newtran error. My config file is as follows: <snip> route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); return; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); return; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { record_route_preset("60.000.000.000:5060;nat=yes"); # insert IP address } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { setflag(1); end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(5); route(1); return; }; if (uri==myself) { if (method=="ACK") { route(6); return; } else if (method=="CANCEL") { route(1); return; } else if (method=="INVITE") { route(3); return; } else if (method=="REGISTER") { route(2); return; }; lookup("aliases"); if (uri!=myself) { route(5); route(1); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; }; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); log(1,"route[2] (Register handler"); if (!search("^Contact: \*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); return; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); return; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (!allow_trusted() && !is_uri_host_local()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); }; lookup("aliases"); if (uri!=myself) { route(5); route(1); return; }; if (uri=~"^sip:[+|00|011][0-9]*@") { # International PSTN log(1,"international PSTN match"); route(4); return; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{11}@") { # Domestic(US) PSTN # route(4); return; }; sl_send_reply("404", "User Not Found"); return; }; if (method=="CANCEL") { route(1); return; }; setflag(1); setflag(3); route(5); route(1); } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- #UAC module routing# t_on_failure("1"); resetflag(9); log(1,"resetflag(9)"); t_relay_to_udp("203.100.100.100","5060"); #PSTN gateway log(1,"host rewritten"); avp_write("i:45", "inv_timeout"); setflag(1); setflag(3); route(5); route(1); } route[5] { # ----------------------------------------------------------------- # RTP Proxy Enabler # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; } route[6] { # ------------------------------------------------------------------------ # ACK Handler # ------------------------------------------------------------------------ # ------------------------------------------------------------------------ # Aliases Section # ------------------------------------------------------------------------ lookup("aliases"); if (uri!=myself) { route(5); route(1); return; }; lookup("location"); route(1); } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:\ 0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } failure_route[1] { log(1,"hit failure_route[1]"); if (t_check_status("401|407")) { log(1,"status is 401 or 407"); if (isflagset(9)) { log(1,"flag9 is set"); t_reply("503","Authentication failed"); return; } if (uac_auth()) { log(1, "called uac_auth"); setflag(9); t_on_failure("1"); append_branch(); t_relay(); } } } The SIP traffic is as follows: U 192.168.0.50:5060 -> 192.168.0.6:5060 INVITE sip:021555222@domain.co.nz SIP/2.0..Via: SIP/2.0/UDP 192.168.0.5 0:5060;rport;branch=z9hG4bK29BECBA8F97C4648A1E027E5DF93BEC0..From: Cameron local Domain <sip:6494466546@domain.co.nz>;tag=3738281931..To: <sip :021555222@domain.co.nz>..Contact: <sip:6494466546@192.168.0.50:5060>.. Call-ID: B1402DE5-6390-4715-A03F-9B1F78AB60CF@192.168.0.50..CSeq: 28000 INV ITE..Proxy-Authorization: Digest username="6494466546",realm="domain.co .nz",nonce="44329b8ecd24c22c48293896310ed7d403738b94",response="014bc155e1a 3554039f6688e9d1e8d42",uri="sip:021555222@domain.co.nz"..Max-Forwards: 70..Content-Type: application/sdp..User-Agent: X-Lite release 1103m..Conten t-Length: 301....v=0..o=6494466546 766866495 766868168 IN IP4 192.168.0.50. .s=X-Lite..c=IN IP4 192.168.0.50..t=0 0..m=audio 8000 RTP/AVP 0 8 3 98 97 1 01..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpm ap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000. .a=fmtp:101 0-15.. #### U 192.168.0.6:5060 -> 192.168.0.50:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 192.16 8.0.50:5060;rport=5060;branch=z9hG4bK29BECBA8F97C4648A1E027E5DF93BEC0..From : Cameron local Domain <sip:6494466546@domain.co.nz>;tag=3738281931 ..To: <sip:021555222@domain.co.nz>..Call-ID: B1402DE5-6390-4715-A03F-9B 1F78AB60CF@192.168.0.50..CSeq: 28000 INVITE..Server: OpenSer (1.0.1 (i386/l inux))..Content-Length: 0..Warning: 392 192.168.0.6:5060 "Noisy feedback te lls: pid=8125 req_src_ip=192.168.0.50 req_src_port=5060 in_uri=sip:0214462 02(a)domain.co.nz out_uri=sip:021555222@domain.co.nz via_cnt==1".... # U 192.168.0.6:5060 -> 203.184.16.2:5060 INVITE sip:021555222@domain.co.nz SIP/2.0..Record-Route: <sip:60.234.13 7.50:5060;nat=yes;ftag=3738281931;lr=on>..Via: SIP/2.0/UDP 192.168.0.6;bran ch=z9hG4bKbdfc.69423e86.0..Via: SIP/2.0/UDP 192.168.0.50:5060;rport=5060;br anch=z9hG4bK29BECBA8F97C4648A1E027E5DF93BEC0..From: Cameron local domain <sip:6494466546@domain.co.nz>;tag=3738281931..To: <sip:021555222@domain.co.nz>..Contact: <sip:6494466546@192.168.0.50:5060>..Call-ID: B1402D E5-6390-4715-A03F-9B1F78AB60CF@192.168.0.50..CSeq: 28000 INVITE..Max-Forwar ds: 69..Content-Type: application/sdp..User-Agent: X-Lite release 1103m..Co ntent-Length: 301....v=0..o=6494466546 766866495 766868168 IN IP4 192.168.0 .50..s=X-Lite..c=IN IP4 192.168.0.50..t=0 0..m=audio 8000 RTP/AVP 0 8 3 98 97 101..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a= rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8 000..a=fmtp:101 0-15.. #### U 192.168.0.6:5060 -> 192.168.0.50:5060 SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL)..Via: SIP/2.0/ UDP 192.168.0.50:5060;rport=5060;branch=z9hG4bK29BECBA8F97C4648A1E027E5DF93 BEC0..From: Cameron local Domain <sip:6494466546@domain.co.nz>;tag= 3738281931..To: <sip:021555222@domain.co.nz>;tag=329cfeaa6ded039da25ff8 cbb8668bd2.bcdf..Call-ID: B1402DE5-6390-4715-A03F-9B1F78AB60CF(a)192.168.0.50 ..CSeq: 28000 INVITE..Server: OpenSer (1.0.1 (i386/linux))..Content-Length: 0..Warning: 392 192.168.0.6:5060 "Noisy feedback tells: pid=8125 req_src_ ip=192.168.0.50 req_src_port=5060 in_uri=sip:021555222@domain.co.nz out _uri=sip:021555222@domain.co.nz via_cnt==1".... # Any help would be appreciated Cameron

18 years, 1 month

[Serusers] Contact header during INVITE

by PREVOT Paul

Hello, I have an issue with the fix_nated_contact() function. OpenSer is supposed to change Contact header, but it only changes it in the INVITE message, but not in the OK response. As a consequence, ACK messages contain sip:test@192.168.1.1:5060 and not the public ip. OpenSer doesn't know how to handle that ACK and forward it to 192.168.1.1:5060. The same issue occurs in CANCEL/487. Changes are done in CANCEL Contact header, but not in 487 reply. No problem for BYE, as OK response doesn't need an ACK I suppose I have something wrong in my cfg file, but I can't find it out. Regards, Paul Openser.Cfg: ...default loadmodules and modparam ... # 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"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # 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(); force_rport(); fix_nated_register(); fix_nated_contact(); }; #if(search("Contact")){ #setflag(6); #force_rport(); # fix_nated_contact(); #}; if(method!="REGISTER" && method!="ACK") { #record everything besides registers and acks setflag(1); setflag(3); }; # 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); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay_to_tls("IP_domain1","port_domain1"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay_to_tls("IP_domain2","port_domain2"); # exit; #} route(1); }; # 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") { #change contact field IP and port by those received really used after NAT rewritting by router #fix_nated_contact(); #force openSer to respond on the port from which the request has been sent insteed of the port #in contact field. force_rport(); # Uncomment this if you want to use digest authentication if (!www_authorize("mydomain", "subscriber")) { www_challenge("mydomain ", "0"); exit; }; setflag(1); setflag(3); fix_nated_register(); save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; 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(); }; exit; }

18 years, 1 month

[Users] Registering and rtpproxy/mediaproxy

by Jeff Williams

We have a commercial SIP server which currently does our registering, low cost routing and customer billing. However, it doesn't handle NATed connections well. To fix this, I wanted to use openser in front of it for customer connections using mediaproxy/rtpproxy. However, I have a problem with this setup. >From what I gather, for the proxy to work properly, the end user need to be registered with the proxy so that it can do nat pings and keep the nat connection alive. But for users to make calls using our commercial server (which is what we want), they must be registered there. Is it possible (or even a good idea) to have a user registered on 2 servers at the same time? How else can we do this? This is what I want to achieve: - from the end user perspective, they register and initiate calls only via the openser proxy. They know nothing of our commercial server. - calls between users is done on the openser proxy - calls to external destinations (i.e. PSTN) go via our commercial server and require that the user is registered there Is this possible? Jeff

18 years, 1 month

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sr-users April 2006