sr-users August 2007

sr-users@lists.kamailio.org

169 participants
286 discussions

by dciepiel＠orange.pl

It comes from Asterisk with max_forward 70. Question is why is it changed (and not to 69 value)? cheers, Daniel ---Oryginalna wiadomość--- Nadawca:"Greger V. Teigre" <greger(a)teigre.com> Do:dciepiel@orange.pl Otrzymana:2007-08-10 10:54 Temat:Re: [Serusers] Why max_forwards=16? SER is forwarding the INVITE, look at where it comes from. g-) dciepiel(a)orange.pl wrote: > Ser is isending INVITE message to Xlite, with max forwards 16 / > instead of 70? Why? > > > cheers, > > Daniel > > ------------------------------------------------------------------------ > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

17 years, 3 months

[Serusers] Need Help!!!

by Kapil Goel

Hi, I want to make an application server using SER with few more entities. The system would be like this | XLite Phone | -------------------- | SIP Proxy Server | --------------------- | App Server using SER | -------------------------- | Media Server | Now these are the tasks I need to accomplish: 1. Registering App Server with SIP Proxy Server. (How to register SER with another Proxy Server?) 2. It should also behave like a proxy. (SER should forward the packet to and from Media Server) 3. RTP must only be created b/w XLite Phone and Media Server. 4. And finally sending INFO Method from App Server to Media Server with MSCML Script. (How does SER send INFO Method with an xml script as a body?) Please suggest something how should I achieve it. Thanks and Regards, Kapil Goel

17 years, 3 months

[Serusers] ODP:Re: Missing ACK and BYE in SIP communication.

by dciepiel＠orange.pl

Yes, I have fixed it. The record_route parameter was set up on local IP. cheers, Daniel ---Oryginalna wiadomość--- Nadawca:"Greger V. Teigre" <greger(a)teigre.com> Do:dciepiel@orange.pl Otrzymana:2007-08-10 10:59 Temat:Re: [Serusers] Missing ACK and BYE in SIP communication. Make sure your record routing and Contact fixing is correct. g-) dciepiel(a)orange.pl wrote: > Hi, > > > I have a short question about ACK. Im my case it is not sent from > callking side (X-lite). Why is it like that? How does Xlite know, not > to send ACK? And why there is no BYE? See flow above. > > Flow: > > > No. Time Source Destination > Protocol Info > 45 6.979169 137.138.137.166 192.168.2.2 > SIP/SDP Request: INVITE sip:6891@wilanowska.homeip.net, with session > description > 46 6.979736 192.168.2.2 137.138.137.166 SIP > Status: 100 trying -- your call is important to us > 47 6.979950 192.168.2.2 81.18.210.159 > SIP/SDP Request: INVITE > sip:6891@81.18.210.159:10020;rinstance=18352dcff8f568bc, with session > description > 51 7.241103 192.168.2.2 81.18.210.159 > SIP/SDP Request: INVITE > sip:6891@81.18.210.159:10020;rinstance=18352dcff8f568bc, with session > description > 52 7.286054 81.18.210.159 192.168.2.2 SIP > Status: 180 Ringing > 53 7.286401 192.168.2.2 137.138.137.166 SIP > Status: 180 Ringing > 54 7.320840 81.18.210.159 192.168.2.2 SIP > Status: 180 Ringing > 55 7.321229 192.168.2.2 137.138.137.166 SIP > Status: 180 Ringing > 102 14.357178 81.18.210.159 192.168.2.2 > SIP/SDP Status: 200 OK, with session description > 103 14.357744 192.168.2.2 137.138.137.166 > SIP/SDP Status: 200 OK, with session description > 104 14.955213 81.18.210.159 192.168.2.2 > SIP/SDP Status: 200 OK, with session description > 105 14.955567 192.168.2.2 137.138.137.166 > SIP/SDP Status: 200 OK, with session description > 109 15.960539 81.18.210.159 192.168.2.2 > SIP/SDP Status: 200 OK, with session description > 110 15.961118 192.168.2.2 137.138.137.166 > SIP/SDP Status: 200 OK, with session description > 117 17.968614 81.18.210.159 192.168.2.2 > SIP/SDP Status: 200 OK, with session description > 118 17.969011 192.168.2.2 137.138.137.166 > SIP/SDP Status: 200 OK, with session description > 135 21.982902 81.18.210.159 192.168.2.2 > SIP/SDP Status: 200 OK, with session description > 136 21.983229 192.168.2.2 137.138.137.166 > SIP/SDP Status: 200 OK, with session description > 171 25.999316 81.18.210.159 192.168.2.2 > SIP/SDP Status: 200 OK, with session description > 172 25.999641 192.168.2.2 137.138.137.166 > SIP/SDP Status: 200 OK, with session description > 193 30.018608 81.18.210.159 192.168.2.2 > SIP/SDP Status: 200 OK, with session description > 194 30.018936 192.168.2.2 137.138.137.166 > SIP/SDP Status: 200 OK, with session description > 229 34.031607 81.18.210.159 192.168.2.2 > SIP/SDP Status: 200 OK, with session description > 230 34.032134 192.168.2.2 137.138.137.166 > SIP/SDP Status: 200 OK, with session description > 251 38.053494 81.18.210.159 192.168.2.2 > SIP/SDP Status: 200 OK, with session description > 252 38.054124 192.168.2.2 137.138.137.166 > SIP/SDP Status: 200 OK, with session description > 269 42.070209 81.18.210.159 192.168.2.2 > SIP/SDP Status: 200 OK, with session description > 270 42.070736 192.168.2.2 137.138.137.166 > SIP/SDP Status: 200 OK, with session description > 325 46.088117 81.18.210.159 192.168.2.2 > SIP/SDP Status: 200 OK, with session description > 326 46.088445 192.168.2.2 137.138.137.166 > SIP/SDP Status: 200 OK, with session description > > > cheers, > > Daniel > > > ------------------------------------------------------------------------ > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

17 years, 3 months

[Serusers] Missing ACK and BYE in SIP communication.

by dciepiel＠orange.pl

Hi, I have a short question about ACK. Im my case it is not sent from callking side (X-lite). Why is it like that? How does Xlite know, not to send ACK? And why there is no BYE? See flow above. Flow: No. Time Source Destination Protocol Info 45 6.979169 137.138.137.166 192.168.2.2 SIP/SDP Request: INVITE sip:6891@wilanowska.homeip.net, with session description 46 6.979736 192.168.2.2 137.138.137.166 SIP Status: 100 trying -- your call is important to us 47 6.979950 192.168.2.2 81.18.210.159 SIP/SDP Request: INVITE sip:6891@81.18.210.159:10020;rinstance=18352dcff8f568bc, with session description 51 7.241103 192.168.2.2 81.18.210.159 SIP/SDP Request: INVITE sip:6891@81.18.210.159:10020;rinstance=18352dcff8f568bc, with session description 52 7.286054 81.18.210.159 192.168.2.2 SIP Status: 180 Ringing 53 7.286401 192.168.2.2 137.138.137.166 SIP Status: 180 Ringing 54 7.320840 81.18.210.159 192.168.2.2 SIP Status: 180 Ringing 55 7.321229 192.168.2.2 137.138.137.166 SIP Status: 180 Ringing 102 14.357178 81.18.210.159 192.168.2.2 SIP/SDP Status: 200 OK, with session description 103 14.357744 192.168.2.2 137.138.137.166 SIP/SDP Status: 200 OK, with session description 104 14.955213 81.18.210.159 192.168.2.2 SIP/SDP Status: 200 OK, with session description 105 14.955567 192.168.2.2 137.138.137.166 SIP/SDP Status: 200 OK, with session description 109 15.960539 81.18.210.159 192.168.2.2 SIP/SDP Status: 200 OK, with session description 110 15.961118 192.168.2.2 137.138.137.166 SIP/SDP Status: 200 OK, with session description 117 17.968614 81.18.210.159 192.168.2.2 SIP/SDP Status: 200 OK, with session description 118 17.969011 192.168.2.2 137.138.137.166 SIP/SDP Status: 200 OK, with session description 135 21.982902 81.18.210.159 192.168.2.2 SIP/SDP Status: 200 OK, with session description 136 21.983229 192.168.2.2 137.138.137.166 SIP/SDP Status: 200 OK, with session description 171 25.999316 81.18.210.159 192.168.2.2 SIP/SDP Status: 200 OK, with session description 172 25.999641 192.168.2.2 137.138.137.166 SIP/SDP Status: 200 OK, with session description 193 30.018608 81.18.210.159 192.168.2.2 SIP/SDP Status: 200 OK, with session description 194 30.018936 192.168.2.2 137.138.137.166 SIP/SDP Status: 200 OK, with session description 229 34.031607 81.18.210.159 192.168.2.2 SIP/SDP Status: 200 OK, with session description 230 34.032134 192.168.2.2 137.138.137.166 SIP/SDP Status: 200 OK, with session description 251 38.053494 81.18.210.159 192.168.2.2 SIP/SDP Status: 200 OK, with session description 252 38.054124 192.168.2.2 137.138.137.166 SIP/SDP Status: 200 OK, with session description 269 42.070209 81.18.210.159 192.168.2.2 SIP/SDP Status: 200 OK, with session description 270 42.070736 192.168.2.2 137.138.137.166 SIP/SDP Status: 200 OK, with session description 325 46.088117 81.18.210.159 192.168.2.2 SIP/SDP Status: 200 OK, with session description 326 46.088445 192.168.2.2 137.138.137.166 SIP/SDP Status: 200 OK, with session description cheers, Daniel

17 years, 3 months

[Serusers] Why max_forwards=16?

by dciepiel＠orange.pl

Ser is isending INVITE message to Xlite, with max forwards 16 / instead of 70? Why? cheers, Daniel

17 years, 3 months

[Serusers] "Noisy feedbeck tells" in SIP message Trying.

by dciepiel＠orange.pl

Hi, Does anybody know something about "Noisy feedback tells" warning in SIP trying message. I will be grateful for any help. cheers, Daniel

17 years, 3 months

[OpenSER-Users] Why if (uri==myself) is always true? (about openser call number pstn)

by zze-WU Yin RD-MAPS-LAN

Hello everybody, I want that openser can call a PSTN number for example: 0296487631. But in the request of INVITE, I see that the callee is always 0296487631(a)opensersip.test.machine ( and opensersip.test.machine is my sip server), that means the (uri==myself) is always ture. So this request of INVITE can not reache the gateway of PSTN. Anybody can give me some advices? This is my openser.cfg and I use openser+radius+mysql. Thank you very much!!! Yin # # $Id: openser.cfg 1827 2007-03-12 15:22:53Z bogdan_iancu $ # simple quick-start config script # Please refer to the Core CookBook at http://www.openser.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) children=4 # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes # port=5060 # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- #set module path mpath="/usr/local/lib/openser/modules/" # Uncomment this if you want to use SQL database loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "auth.so" #loadmodule "auth_db.so" loadmodule "auth_radius.so" loadmodule "group_radius.so" loadmodule "acc.so" # ----------------- setting module-specific parameters --------------- # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- modparam("usrloc", "db_mode", 0) modparam("auth_radius|group_radius|acc", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("group_radius","use_domain", 0) modparam("auth_radius", "service_type", 15) modparam("auth", "secret", "testing123") modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) modparam("acc", "service_type", 15) modparam("acc", "radius_extra", "User-Name=$fU;Sip-Src-IP=$si;Sip-Src-Port=$sp") # 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) # ------------------------- 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"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; #-------set flag for Radius Accounting #if (!method=="OPTIONS") setflag(3); #if (method=="INVITE"){ # log (1, "INVITE MESSAGE RECEIVED - START ACC\n"); # setflag(1); # setflag(2); #}; if (method=="BYE"){ log (1, "BYE - STOP ACCOUNTING\n"); setflag(1); }; if (method=="CANCEL"){ log (1, "CANCEL -STOP ACCOUNTING\n"); setflag(1); }; # 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); }; 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("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} # route(1); if (method=="ACK") { route(1); exit; }else if (method=="CANCEL"){ route(1); exit; }else if (method=="INVITE") { route(3); exit; }else if (method=="REGISTER") { route(2); exit; }; lookup("aliases"); if (uri!=myself) { route(1); exit; }; if (!lookup("location")) { sl_send_reply("404","Not User PSTN Found"); 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") { # Uncomment this if you want to use digest authentication if (!radius_www_authorize("opensersip.test.machine")) { www_challenge("opensersip.test.machine", "0"); exit; }; save("location"); exit; }; if (method=="INVITE") { if (!radius_proxy_authorize("opensersip.test.machine")){ proxy_challenge("opensersip.test.machine","0"); 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 SIP User 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; } route[2] { # Register Message Handler sl_send_reply("100","Trying"); if (!radius_www_authorize("opensersip.test.machine")) { www_challenge("opensersip.test.machine", "0"); exit; }; if (!save("location")) { sl_reply_error(); }; } route[3] { # Invite Message Handler # if (!radius_proxy_authorize("opensersip.test.machine")){ # proxy_challenge("opensersip.test.machine","0"); # exit; # } # lookup("aliases"); # if (!uri==myself) { # append_hf("P-hint: outbound alias\r\n"); # route(1); # }; if (uri=~"^sip:[0-9]{10}@") { route(4); exit; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { route(4); exit; }; sl_send_reply("404","User not Found"); exit; }; route(1); } route[4] { rewritehost("172.20.14.32"); route(1); }

17 years, 3 months

[OpenSER-Users] A function to know if a user is registered?

by Iñaki Baz Castillo

Because I use pua_usrloc I want to apply pua_set_publish() in a REGISTER only if the user wasn't already registered (in order to avoid new NOTIFY with just "online" which can substitute the user client real state PUBLISH). So I need a function to determine if the user is registered. I can't use: if ( ! lookup("location") { ... because that function checks the URI in "location" table, and in a REGISTER message the URI is just: sip:server So I need a function to check if the "To:" field ($tu) is registered. Is that exist? or should I make a custom avp_query and so? Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

17 years, 3 months

[Serusers] RPM spec file for ser_ctl

by Bruce Bauman

Before I create one, has anyone written an RPM spec file for the "new" ser_ctl? If not, I'll write/contribute one. Thanks. -- Bruce

17 years, 3 months

[Serusers] SER+PATTON1+PATTON2+PATTON3

by pokerino76

I have 3 patton from 8 channels everyone. I must make the sum of the channels 8+8+8=24. The patton they are connects to with the SER. How to make the switch of the channels when one of the three is saturate? Thanks.

17 years, 3 months

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sr-users August 2007