sr-users March 2005

sr-users@lists.kamailio.org

250 participants
627 discussions

[Serusers] logging calls per hour
by Iqbal 01 Apr '05

01 Apr '05

Hi Anyone know of a simple script to pull calls per hour details out of acc. I just want a no. calls per hour per day details, so I can throw into a graph. Iqbal

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[Serusers] How Do I Check ACLs When Caller ID Blocking Is Enabled?
by Java Rockx 31 Mar '05

31 Mar '05

Hi All. How do I effectively use is_user_in() to check group permissions when the SIP client has caller-id blocking enabled? In other words, how can I determine if a caller has "free-pstn" access when they enable caller-id blocking on their SIP phone? Here is what I'm doing: if (is_user_in("From", "free-pstn")) { setflag(30); }; Regards, Paul

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Re: [Serusers] how to get phone number
by info＠beeplove.com 31 Mar '05

31 Mar '05

Iqbal, I am from USA. Java Rockx, Thanks for your help. I might go with Asterisk. Thanks, Mohammad Original Message: ----------------- From: Iqbal iqbal(a)gigo.co.uk Date: Thu, 31 Mar 2005 19:29:58 +0100 To: info(a)beeplove.com, serusers(a)iptel.org Subject: Re: [Serusers] how to get phone number which country you in info(a)beeplove.com wrote: >Hello: > >This is not a SER related topics. > >How can I get a phone number (real phone number) for my SIP device? > > >Thanks, >Mohammad > >-------------------------------------------------------------------- >mail2web - Check your email from the web at >http://mail2web.com/ . > > >_______________________________________________ >Serusers mailing list >Serusers(a)iptel.org >http://mail.iptel.org/mailman/listinfo/serusers > >. > > > -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

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Re: [Serusers] how to get phone number
by info＠beeplove.com 31 Mar '05

31 Mar '05

Other than, CiscoAS5300 what are other PSTN gateway available? Thanks, Mohammad Original Message: ----------------- From: Java Rockx javarockx(a)gmail.com Date: Thu, 31 Mar 2005 13:14:36 -0500 To: info(a)beeplove.com, serusers(a)iptel.org Subject: Re: [Serusers] how to get phone number If you have a PSTN gateway such as a Cisco AS5300 then the PRI will have DIDs (phone numbers) which you can assign to SIP users. Anyhow, the trick here is to get a PRI from your local phone company. Regards, Paul On Thu, 31 Mar 2005 12:44:22 -0500, info(a)beeplove.com <info(a)beeplove.com> wrote: > Hello: > > This is not a SER related topics. > > How can I get a phone number (real phone number) for my SIP device? > > Thanks, > Mohammad > > -------------------------------------------------------------------- > mail2web - Check your email from the web at > http://mail2web.com/ . > > _______________________________________________ > Serusers mailing list > Serusers(a)iptel.org > http://mail.iptel.org/mailman/listinfo/serusers > -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

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[Serusers] how to get phone number
by info＠beeplove.com 31 Mar '05

31 Mar '05

Hello: This is not a SER related topics. How can I get a phone number (real phone number) for my SIP device? Thanks, Mohammad -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

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[Serusers] Nathelper/RTPProxy not working for agents behind NAT
by Vivienne Curran 31 Mar '05

31 Mar '05

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[Serusers] Nathelper/RTPProxu not working for user agents behind NAT
by Vivienne Curran 31 Mar '05

31 Mar '05

Hi, I am having problems troubleshooting a problem I am experiencing with my SER configuration. I have ser 0.8.14 running with rtpproxy and nathelper enabled. I have two phones on the same subnet behind nat and I would like to make a call between the two. I want to invoke rtpproxy for this as they both have private address [I know this isnt the most efficient way as theyre both on the same subnet but I can worry about that later]. When I ring from the phone 1 (2092) to phone 2 (2093), 2092 can hear voice but 2093 cant. When 2093 ring 2092, theres no audio. These phones are Grandstream BT100s. They have been configured to listen on different SIP and RTP ports. 2092: SIP Port: 5060 2092: RTP Port: 5004 2093: SIP Port: 5061 2093: RTP Port: 5005 I have included my ser.cfg file below. I can send on the SER SIP message sniff if nescessary [I tried attaching it but serbouncers said the attachment was too big]. I can confirm that my rtpproxy is working (originally I thought it wasnt) by using strace d <rtpproxy pid> -f F. I can see a signal being returned. Any help would be appreciated or advise as to how I can proceed troubleshooting. Kindest Regards, Vivienne. # # $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" alias="84.203.148.146:5060" # ------------------ 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/nathelper.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_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", 11) #!! Nathelper modparam("registrar", "nat_flag", 60) modparam("nathelper", "natping_interval", 10) #Ping interval 10 s modparam("nathelper", "ping_nated_only", 1) #Ping only clients behind NAT modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock") modparam("tm", "fr_inv_timer", 20) # ------------------------- 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; }; if (nat_uac_test("3")){ if (method == "REGISTER" || ! search("^Record-Route:")){ log("Log: Someone trying to register from private IP,rew riting\n"); fix_nated_contact(); #Rewrite contact with source IP if (method == "INVITE"){ fix_nated_sdp("1"); #Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as Nated }; }; ################################################################### # 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(); # loose-route processing if (loose_route()) { #commented 11/02/05 #t_relay(); 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) { log(1,"into loop"); if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("84.203.148.146", "subscriber")) { # www_challenge("84.203.148.146", "0"); # break; # }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; if (method=="INVITE"){ log(1,"in invite loop"); #break; #no 100 trying t_on_failure("1"); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { #sl_send_reply("404", "Not Found"); route(2); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP #commented 11/02/05####################### if (!t_relay()) { sl_reply_error(); }; #route(1); } ################################## route[1] { #!!Nathelper 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 IP addresses") ; break; }; if (isflagset(6)){ force_rtp_proxy(); } t_on_reply("1"); if(!t_relay()){ sl_reply_error(); break; }; } ###################################### #!! Nathelper onreply_route[1] { if(isflagset(6) && status =~ "(183)|2[0-9][0-9]"){ fix_nated_contact(); force_rtp_proxy(); }else if (nat_uac_test("1")){ fix_nated_contact(); }; } # ------------- handling of unavailable user ------------------# route[2] { # non-Voip -- just send "off-line" if (!(method == "INVITE" || method == "ACK" || method == "CANCEL")) { sl_send_reply("404", "Not Found"); break; }; # forward to voicemail now rewritehostport("84.203.148.146:5062"); t_relay_to_udp("84.203.148.146", "5062"); } # if forwarding downstream did not succeed, try voicemail running # at 84.203.148.146:5062 failure_route[1] { revert_uri(); rewritehostport("84.203.148.146:5062"); append_branch(); t_relay_to_udp("84.203.148.146", "5062"); } Send instant messages to your online friends http://uk.messenger.yahoo.com

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[Serusers] SER --> RADIUS + MYSQL - Urgent problem !
by a cairo 31 Mar '05

31 Mar '05

Dear list, i have installed SER with radiusclient on a server and RADIUS (freeradius) with MYSQL DataBase on another server. SerRadius HOWTO is referred to a configuration on the same machine and with digest authentication. In my situation user are into mysql database and not in a digest file. At this point i followed instructions disabling "digest" word in authorize() and autenticathed() section in freeradius.conf. Tested with a simple client my server radius with mysql, response are fast and correct but when i try with SER i note that password is note received from radius! Is radiusclient the problem? Work radiusclient with "local" password or only with digest authentication? If digest authentication is request, it's impossible create a digest file for each user !!! Can anyone help me? Thanks Arcibald

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[Serusers] How to add a P-Asserted-Identity header ?
by Lucas Aimaretto 31 Mar '05

31 Mar '05

That's it ... how to add that header? I believe is using the append_hf, but do not know which parameters should be used ... Regards, Lucas -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.6 - Release Date: 30/03/2005

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[Serusers] Nathelper/RTPPorxy not working for two natted clients
by Vivienne Curran 31 Mar '05

31 Mar '05

Hi, I am having problems troubleshooting a problem I am experiencing with my SER configuration. I have ser 0.8.14 running with rtpproxy and nathelper enabled. I have two phones on the same subnet behind nat and I would like to make a call between the two. I want to invoke rtpproxy for this as they both have private address [I know this isnt the most efficient way as theyre both on the same subnet but I can worry about that later]. When I ring from the phone 1 (2092) to phone 2 (2093), 2092 can hear voice but 2093 cant. When 2093 ring 2092, theres no audio. These phones are Grandstream BT100s. They have been configured to listen on different SIP and RTP ports. 2092: SIP Port: 5060 2092: RTP Port: 5004 2093: SIP Port: 5061 2093: RTP Port: 5005 I have included my ser.cfg file below. I can send on the SER SIP message sniff if nescessary [I tried attaching it but serbouncers said the attachment was too big]. I can confirm that my rtpproxy is working (originally I thought it wasnt) by using strace d <rtpproxy pid> -f F. I can see a signal being returned. Any help would be appreciated or advise as to how I can proceed troubleshooting. Kindest Regards, Vivienne. # # $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" alias="84.203.148.146:5060" # ------------------ 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/nathelper.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_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", 11) #!! Nathelper modparam("registrar", "nat_flag", 60) modparam("nathelper", "natping_interval", 10) #Ping interval 10 s modparam("nathelper", "ping_nated_only", 1) #Ping only clients behind NAT modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock") modparam("tm", "fr_inv_timer", 20) # ------------------------- 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; }; if (nat_uac_test("3")){ if (method == "REGISTER" || ! search("^Record-Route:")){ log("Log: Someone trying to register from private IP,rew riting\n"); fix_nated_contact(); #Rewrite contact with source IP if (method == "INVITE"){ fix_nated_sdp("1"); #Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as Nated }; }; ################################################################### # 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(); # loose-route processing if (loose_route()) { #commented 11/02/05 #t_relay(); 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) { log(1,"into loop"); if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("84.203.148.146", "subscriber")) { # www_challenge("84.203.148.146", "0"); # break; # }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; if (method=="INVITE"){ log(1,"in invite loop"); #break; #no 100 trying t_on_failure("1"); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { #sl_send_reply("404", "Not Found"); route(2); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP #commented 11/02/05####################### if (!t_relay()) { sl_reply_error(); }; #route(1); } ################################## route[1] { #!!Nathelper 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 IP addresses") ; break; }; if (isflagset(6)){ force_rtp_proxy(); } t_on_reply("1"); if(!t_relay()){ sl_reply_error(); break; }; } ###################################### #!! Nathelper onreply_route[1] { if(isflagset(6) && status =~ "(183)|2[0-9][0-9]"){ fix_nated_contact(); force_rtp_proxy(); }else if (nat_uac_test("1")){ fix_nated_contact(); }; } ######## # ------------- handling of unavailable user ------------------ route[2] { # non-Voip -- just send "off-line" if (!(method == "INVITE" || method == "ACK" || method == "CANCEL")) { sl_send_reply("404", "Not Found"); break; }; # forward to voicemail now rewritehostport("84.203.148.146:5062"); t_relay_to_udp("84.203.148.146", "5062"); } # if forwarding downstream did not succeed, try voicemail running # at 84.203.148.146:5062 failure_route[1] { revert_uri(); rewritehostport("84.203.148.146:5062"); append_branch(); t_relay_to_udp("84.203.148.146", "5062"); } Send instant messages to your online friends http://uk.messenger.yahoo.com

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sr-users March 2005