sr-users May 2004

sr-users@lists.kamailio.org

154 participants
333 discussions

[Serusers] debug sip message
by Richard 26 May '04

26 May '04

Hi, Is there a way to print the whole sip messages in ser debug mode? I can have another terminal to ngrep the packets, but it is kind of hard to correlate the ngrep results with ser debug. Currently I hack into the first function I call in the config script and add a print. Anyone knows other ways? Thanks, Richard __________________________________ Do you Yahoo!? Friends. Fun. Try the all-new Yahoo! Messenger. http://messenger.yahoo.com/

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[Serusers] Setup Ser, Please help
by Peter Liu 26 May '04

26 May '04

I am try to learn set up SER right now. I really need someone who can help me with my case: I want to set up my own Ser in my home computer and I want to let people from different regions use SER in my home computer. I have read HOWTO file, but I don't know how to setup SIP_DOMAIN variable and how to configure DNS SVR records(I don't have DNS server). DO I really need to have DNS server? SIP_DOMAIN is local domain or I need to register one? Thanks, Peter __________________________________ Do you Yahoo!? Friends. Fun. Try the all-new Yahoo! Messenger. http://messenger.yahoo.com/

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[Serusers] PSTN to SIP in analog gateways
by Humberto Atristain 26 May '04

26 May '04

Dear Sirs I have a very important Issue (for me) , how do I automatically send a call from an analog gateway to a defined IP user For me the PSTN termination is easy but what about inbound calling, I need no IVR I need only to send to a determined sip user. The gateway has 2 options Peer-to-peer (I can point to calls toa n IP and a e164 number) gonder if this can be a sip user) And Proxy registration What I need to use? Please help HA

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[Serusers] forward to PSTN
by Ri Liang 25 May '04

25 May '04

Please adivise if there any variable that i can get some data from the MySql. My problem is as follow: When a particular user (for example 123(a)foo.bar) need to call to PSTN, we need to sent in the "From" part as 456(a)foo.bar, because the PSTN provider needs the 456 to authenticate. so we need some way to transform the "From" part of the user from 123(a)foo.bar ----> 456(a)foo.bar before forward the call to PSTN. I think that it may be someting like (of course, the lookup2() is what i am looking for) ...... if ( PSTN_pin = lookup2 ("pin_table") ) { subst('/^From:(.*)sip:[^@]*@[a-zA-Z0-9.]+(.*)$/From:\1sip:PSTN_pin@pstn.foo. bar\2/g'); forward( pstn.foo.bar, 5060 ); break; } ........ Thanks Ri Liang

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[Serusers] SER parallel forking
by Peter Gradwell 25 May '04

25 May '04

Hi, We're integrating SER and Asterisk and want to use parallel forking so that calls for sip users to simultaneously call the sip phone and ser<user>@asterisk.gradwell.net, which will then wait a per-user delay before answering with the voicemail. The call does divert, but only after SER's own timer expires. This might work ok as a last resort, but really we want the parallel forking as described above to work so that we can have per-user delays, it may also make supporting call diverts easier, since Asterisk is good at stuff like that. As far as I can tell the main logic for doing the branching (using append_branch) is very similar to SER's example config file onr.cfg in the distribution's examples directory. However I didn't have much luck with that either! Our config file is included below. Any thoughts on how to make this work would be most welcome! many thanks peter ===================================================================== # # $Id: ser.cfg,v 1.21.2.1 2003/07/30 16:46:18 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=9 fork=no log_stderror=yes check_via=no # (cmd. line: -v) dns=yes # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=193.111.200.106 #mhomed=1 port=5060 children=4 fifo="/tmp/ser_fifo" alias="ser.gradwell.net" alias="193.111.200.106" #fifo_db_url="mysql://ser:********@hostingdb/ser" # ------------------ 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/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/uri.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" #loadmodule "/usr/local/lib/ser/modules/enum.so" loadmodule "/usr/local/lib/ser/modules/msilo.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" loadmodule "/usr/local/lib/ser/modules/xlog.so" # Use a private ENUM space [not yet, we just pretend to be the real thing #modparam("enum","domain_suffix","enum.go-sip.org.") # We force all the lookup and registrar related stuff to take the domain # into account. # Point everything at sip-auth-adm for DB related stuff modparam("usrloc", "use_domain", 1) modparam("registrar", "use_domain", 1) modparam("group", "use_domain", 1) modparam("auth_db", "use_rpid", 1) modparam("auth_db", "rpid_column", "username") # We do not use persistant storage, this reduces DB overhead, # however, if we move to a HA pair, then this should be set to 1 # and the seed for generating nonce values must be synchronised. # NOTE: Actually we HAVE to use 1 anyway as our aliases table is # in SQL. modparam("usrloc", "db_mode", 2) modparam("usrloc","db_url", "mysql://ser:********@hostingdb/ser") modparam("domain","db_url", "mysql://ser:********@hostingdb/ser") modparam("domain","db_mode",1) modparam("group","db_url", "mysql://ser:********@hostingdb/ser") #modparam("group","db_mode",1) modparam("auth_db","db_url", "mysql://ser:********@hostingdb/ser") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") #modparam("uri","db_url", "mysql://ser:********@hostingdb/ser") #modparam("uri","use_uri_table", yes) modparam("acc","db_url", "mysql://ser:********@hostingdb/ser") modparam("msilo","db_url","mysql://ser:********@hostingdb/ser") #modparam("msilo","registrar","sip:registrar@go-sip.com") modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "fr_timer", 10 ) #modparam("tm", "wt_timer", 2 ) # Useful for some badly behaved clients modparam("rr", "enable_full_lr", 1) # Set accounting flags, these are the defaults anyway modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) # Nathelpher #modparam("nathelper", "natping_interval", 10) # Which flags mean what... # 1 - account # 2 - missed call # 3 - url reqires enum rewrite # 4 - user has voicemail accessuser has voicemail access # 5 - user is online # 6 - inbout call rtp stream should be proxied: # 7 - outbound call rtp stream should be proxied: # 8 - set up voicemail in route[3] # ------------------------- request routing logic ------------------- # main routing logic route{ xdbg("*****\n"); xdbg("***** %rm %ru\n"); xdbg("*****\n"); # 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(method=="BYE"){ setflag(1); 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 ( is_uri_host_local() || uri == myself ) { # We want to deal primarily with numbers rather # than usernames, this makes life easier in # voicemail. # # Translate any usernames to numbers if (method=="REGISTER") { # Leaving the domain blank means that the # domain from the url will be used. if (!www_authorize("","subscriber")) { www_challenge("", "1"); break; }; save("location"); break; }; log("**** lookup(aliases)\n"); lookup("aliases"); # We don't deal with presence at the moment if (method=="SUBSCRIBE" || method == "PUBLISH") { sl_send_reply("503", "Service Unavailable"); break; }; # Rewriting for other SIP networks... if(uri=~"^sip:\*\*.*"){ # We only want our customer relaying through our servers strip(2); if (!proxy_authorize("","subscriber")) { proxy_challenge("", "1"); break; }; sl_send_reply("404", "Not Found"); break; }; if(uri=~"^sip:[0-9][0-9][0-9]@.*" ){ t_relay_to_udp("asterisk.gradwell.net","5060"); break; }; if (uri=~"^sip:0.*") { if (uri=~"^sip:0.*") { if (uri=~"^sip:00.*") { strip(2); }else{ strip(1); prefix("44"); }; }; route(6); break; }; # native SIP destinations are handled using our USRLOC DB if (method == "INVITE" || method == "ACK" || method == "MESSAGE") { if (uri=~"^sip:\*") { t_relay_to_udp("asterisk.gradwell.net","5060"); break; }; # Request uri is now invalid or in Username form if (lookup("location")) { log("*** found in usrloc\n"); if(method=="MESSAGE"){ # Remote agent may not accept messages, we juststore them. t_on_failure("1"); t_relay(); break; }; # they are online setflag(5); } else { if(method=="MESSAGE"){ # Remote agent may not accept messages, we juststore them. if (m_store("1")) { t_reply("202", "Accepted"); }else{ t_reply("503", "Service Unavailable"); }; break; }; }; if (!isflagset(5)) { log("**** not found in usrloc, diverting to vm\n"); revert_uri(); lookup("aliases"); # User not registered with either username or extension # Instant Unavailable voicemail acc_db_request("Unavailable - Offline", "missed_calls"); log("**** lookup(aliases)\n"); lookup("aliases"); prefix("ser"); route(2); break; }; setflag(8); route(3); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP route(3); } # Our voicemail route. route[2]{ append_hf("P-hint: in-route-2\r\n"); log("IDESK: Route 2, Forwarding to Voicemail\n"); xdbg("IDESK: method=%rm, r_uri=%ru, cseq=%cs\n"); rewritehost("asterisk.gradwell.net"); rewriteport("5060"); if (!t_relay()) { log("Forwarding to Voicemail FAILED\n"); sl_reply_error(); break; }; break; } # Stateful relaying with NAT if it is needed. # NOTE: One possibel enhancement here is to do the rtp-proxying via another box # this would move the traffic away from the ser server completely. # Nat for outbound from idesk route[3]{ log("IDESK: Route for fixing up outbound\n"); if (isflagset(8)) { # We know where they are, and they have # voicemail access, so we fork to their # voicemail account. append_branch(); revert_uri(); log("**** setting up vm branch\n"); log("**** lookup(aliases)\n"); lookup("aliases"); prefix("ser"); append_hf("P-hint: known-vm \r\n"); rewritehost("asterisk.gradwell.net"); rewriteport("5060"); }; if (method == "INVITE"){ if (isflagset(6)) { log("IDESK: Outbound RTP Proxying \n"); log("failure 1, reply 1\n"); t_on_failure("1"); #t_on_reply("1"); } ; if (isflagset(7)){ log("IDESK: Inbound RTP Proxying \n"); log("failure 1, reply 2\n"); t_on_failure("1"); #t_on_reply("2"); } ; if (isflagset(5) && !isflagset(6) && !isflagset(7)){ log("failure 1, reply 3\n"); t_on_failure("1"); #t_on_reply("3"); }; }; if (!t_relay()) { sl_reply_error(); break; }; } #### ####onreply_route[1]{ #### # Our rtp proxying #### log("IDESK: Reply Route for fixing up Outboudn nat\n"); #### if(status=~"4[0-9][0-9].*"){ #### setflag(2); #### }; #### if(status=~"200.*" && search("application/sdp")){ #### if(src_ip=="192.168.254.27"){ #### route(5); #### }; #### # If it comes back through this route then it needs natting #### if (search("application/sdp")){ #### force_rtp_proxy_from("192.168.254.26"); #### }; #### }; ####} #### ####onreply_route[2]{ #### # Our rtp proxying #### log("IDESK: Reply Route for fixing up Inbound nat\n"); #### if(status=~"4[0-9][0-9].*"){ #### setflag(2); #### }; #### if(status=~"200.*" && search("application/sdp")){ #### if(src_ip=="194.130.117.27"){ #### route(5); #### }; #### # If it comes back through this route then it needs natting #### if (search("application/sdp")){ #### force_rtp_proxy_from("194.130.117.26"); #### }; #### }; ####} onreply_route[3]{ # Our rtp proxying log("IDESK: Reply Route for fixing up Inbound nat\n"); if(status=~"4[0-9][0-9].*"){ setflag(2); }; if(status=~"200.*" && search("application/sdp")){ if(isflagset(5) && ( src_ip == "192.168.254.27" || src_ip == "194.130.117.27")){ route(5); }; }; } failure_route[1]{ if(method=="MESSAGE"){ # Remote agent may not accept messages, we juststore them. if (m_store("1")) { t_reply("202", "Accepted"); }else{ t_reply("503", "Service Unavailable"); }; break; }; append_hf("P-hint: missed \r\n"); setflag(2); append_hf("P-hint: in-reply-route-1\r\n"); log("IDESK: Failure Route 1\n"); } # Account as missed route[5] { log("IDESK: Route 3\n"); append_hf("P-hint: in-route-3\r\n"); append_hf("P-hint: missed \r\n"); # setflag(2); acc_db_request("No Answer", "missed_calls"); } # This is our route out to the PSTN; route[6]{ append_hf("P-hint: in-route-4-pstn\r\n"); # This really is going to be a PSTN call, so we # Need to check the credentials # Leaving the domain blank means that the # domain from the url will be used. if (uri=~"^sip:44.*") { strip(2); prefix("0"); }; if(method=="INVITE"){ # sip.calluk.com will authenticate... #if (!proxy_authorize("","subscriber")) { # proxy_challenge("", "1"); # break; #}; append_rpid_hf(); log("Forwarding to PSTN\n"); setflag(1); rewritehost("sip.calluk.com"); if(!t_relay_to_udp("sip.calluk.com","5060")){ sl_reply_error(); }; break; }; if(method=="CANCEL" || method=="ACK" || method=="BYE"){ setflag(1); if(!t_relay_to_udp("sip.calluk.com","5060")){ sl_reply_error(); }; break; } } ===================================================================== thanks peter -- peter gradwell. gradwell dot com Ltd. http://www.gradwell.com/ -- engineering & hosting services for email, web and voip -- -- http://www.peter.me.uk/ -- http://www.voip.org.uk/ --

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[Serusers] ZyXel P2000w wlan SIP UA's
by mike 25 May '04

25 May '04

Hey! Recently bought a couple of ZyXel P-2000W sip ua's and have had trouble registering with SER. It wouldn't even register to iptel's proxy. Anyone have any hands on experience with the matter? Certainly is a nice phone. Direct IP-IP calls within a private network work fine and do not even require a proxy, but have problems calling out. Ideas anyone? The product link below. http://www.zyxel.com/product/model.php?indexcate=1075688089&indexFlagvalue=… thanks, Michael Seppänen

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Re: [Serusers] Problem in serweb register
by Jan Janak 25 May '04

25 May '04

Yes but is there any error message (usuall a mandatory field left unfilled) ? If not, is there any error message in apache logs ? Please always CC the mailing list. Jan. On 25-05 16:13, sidf@blrcsv wrote: > Thanks you for you reply so quickly. > > After click register button, the register webpage was return back, that is > the same webpage as before click register button. > > ----- Original Message ----- > From: "Jan Janak" <jan(a)iptel.org> > To: "sidf@blrcsv" <sidf(a)blrcsv.china.bell-labs.com> > Cc: <serusers(a)lists.iptel.org> > Sent: Tuesday, May 25, 2004 4:04 PM > Subject: Re: [Serusers] Problem in serweb register > > > > Is there any error message on that page ? > > > > Jan. > > > > On 25-05 12:54, sidf@blrcsv wrote: > > > > > > I met a problem in register new user using SERWEB. When I click the > register button after i fill out the register item, the page return back > itself, not the finish page. > > > Anyone know why? > > > _______________________________________________ > > > Serusers mailing list > > > serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > >

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[Serusers] redirect server
by mohammad mirzaee 25 May '04

25 May '04

HI ALL; MY ser.cfg for redirection to another branch is something like the following: ....... ....... append_branch("sip:1002@192.168.0.1") sl_send_reply("302", "unavailable"); but i got fast busy, ser debug told me that "LOCAL ACK FOUND,, drop it" Somebody told me there is a thread about redirection with a bugfix but i couldnot find it. Plz help me to fix my problem thnax mohammad

2 1

[Serusers] Presence problem
by andayesh 25 May '04

25 May '04

Hi all, I am using rtc1.2 as the UA, everything is working fine, e.g. video conference and IM, but I have some difficulty using PA. I have added this module, but my clients does not inform about their buddies status.

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[Serusers] SERWEB: How i change the confirmation address in the confirmation email to user?
by sidf＠blrcsv 25 May '04

25 May '04

I installed the SERWEB0.8.12. After i register a new user using the register webpage, a confirmation email will be sent to the user's email address. it look like this: Thank you for registering with ...... We are reserving the following SIP address for you: sip:8002@...... To finalize your registration please check the following URL within 24 hours: http://....../serweb/user/reg/confirmation.php?nr=bd77f30ed0c805c6d204e65eae5aa9d5 (If you confirm later you will have to re-register.) My question is , which and how modify the php file in directory /serweb/user/reg , can make me change the registration confirmation address?Thanks.And , I find a bug in SERWEB, default address in the confirmation email is ...../serweb/user/... ,but it is serweb/user_interfacein the serweb package.siduanfeng

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sr-users May 2004