sr-users August 2005

sr-users@lists.kamailio.org

220 participants
558 discussions

[Serusers] SER <==> SEMS fifo permisions problem

by Pavol Segeč

Hi All ! I'm implementing SER, SEMS server, I've configured fifo as communication method between them (folowing install manual). In ser.cfg I've enabled conference part in INVITE message route handling as follows : if (uri=~"sip:100.*@") { if (!t_newtran()){ sl_send_reply("500","Could not create transaction"); break; }; t_reply("100","Trying - just wait a minute !"); # assumes that Sems configuration parameter 'socket_name=' # has been set to /tmp/am_sock if(!t_write_req("/tmp/am_fifo","conference")) { t_reply("500","error contacting sems - in uri check 100"); }; break; }; I can make a call on sip:100.*, but I don't have received any announcement and in /var/log/syslog I see following problems with SEMS and more oftenly with fifo permisions: Aug 1 14:30:09 mail Sems[25848]: Error: (SerClient.cpp)(read_from_fifo)(286): no more retries! Aug 1 14:30:09 mail Sems[25848]: Error: (SerClient.cpp)(read_from_fifo)(287): last error: Success Aug 1 14:30:09 mail Sems[25848]: Error: (SerClient.cpp)(send)(137): while reading Ser's response. Aug 1 14:30:09 mail Sems[25848]: Error: (AmRequest.cpp)(send)(219): while sending request to Ser Aug 1 14:30:15 mail Sems[25852]: Error: (SerClient.cpp)(read_from_fifo)(286): no more retries! Aug 1 14:30:15 mail Sems[25852]: Error: (SerClient.cpp)(read_from_fifo)(287): last error: Success Aug 1 14:30:15 mail Sems[25852]: Error: (SerClient.cpp)(send)(137): while reading Ser's response. Aug 1 14:30:15 mail Sems[25852]: Error: (AmRequest.cpp)(send)(219): while sending request to Ser Aug 1 14:33:28 mail /usr/sbin/ser[25731]: ERROR: open_reply_pipe: open error (/tmp/0000660D1443C4FC): Permission denied Aug 1 14:33:28 mail /usr/sbin/ser[25731]: ERROR: fifo_reply: no reply pipe /tmp/0000660D1443C4FC Thanks palo

18 years, 9 months

[Serusers] can not open dictionary from freeradiusd!

by zhu lizhong

hello, guys: I installed freeradiusd and SER. now, i want to connect the two together. the problme is that when i include dictionary.ser in raddb/dictionary, and run radiusd -X, then the messages came out as follow: ------------------------------------- read_config_files: reading dictionary Errors reading dictionary: dict_init: /usr/local/etc/raddb/dictionary[14]: Couldn't open dictionary " /usr/local/share/freeradius/dictionary": No such file or directory Errors reading radiusd.conf -------------------------------------- any one knows how to solve it, please give me hints. thanks lot! zhu

18 years, 9 months

[Serusers] Problem with digest authentication

by aimable

Hi serusers, I have been using various versions of SER from last year without any problem but recently I made a new installation of OpenSER 0.9.5. Since then I am having problems with digest authentication from some of the phones. I have a bunch of 186 ATAs and Cisco 7940 phones but they cannot register to the server, while all the soft phones can register successfully. The server says Credentials with given realm not found. I tried to change the realm to localhost, the IP address, with no luck. And below is the result of ngrep I tried to grep the messages from the phones and here below is one message from a Cisco 186 ATA which has failed to register ########### Beginning of the capture ################## U PHONEIP:5060 -> SERVERIP:5060 REGISTER sip:SERVERIP SIP/2.0. Via: SIP/2.0/UDP PHONEIP:5060. From: <sip:06090003@SERVERIP;user=phone>;tag=500808430. To: <sip:06090003@SERVERIP;user=phone>. Call-ID: 704382462@PHONEIP. CSeq: 1 REGISTER. Contact: <sip:06090003@PHONEIP:5060;user=phone;transport=udp>;expires=3600. User-Agent: Cisco ATA 186 v2.16.2 ata18x (030829a). Content-Length: 0. # U SERVERIP:5060 -> PHONEIP:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP PHONEIP:5060. From: <sip:06090003@SERVERIP;user=phone>;tag=500808430. To: <sip:06090003@SERVERIP;user=phone>. Call-ID: 704382462@PHONEIP. CSeq: 1 REGISTER. Server: OpenSer (0.9.5 (i386/linux)). Content-Length: 0. Warning: 392 SERVERIP:5060 "Noisy feedback tells: pid=4490 req_src_ip=PHONEIP req_src_port=5060 in_uri=sip:SERVERIP out_uri=sip:SERVERIP via_cnt==1". # U SERVERIP:5060 -> PHONEIP:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP PHONEIP:5060. From: <sip:06090003@SERVERIP;user=phone>;tag=500808430. To: <sip:06090003@SERVERIP;user=phone>;tag=329cfeaa6ded039da25ff8cbb8668bd2.8af0 . Call-ID: 704382462@PHONEIP. CSeq: 1 REGISTER. WWW-Authenticate: Digest realm="talk.artel.rw", nonce="42edb29e1dbcc6fa814dd3396634ed7be68eea56". Server: OpenSer (0.9.5 (i386/linux)). Content-Length: 0. Warning: 392 SERVERIP:5060 "Noisy feedback tells: pid=4490 req_src_ip=PHONEIP req_src_port=5060 in_uri=sip:SERVERIP out_uri=sip:SERVERIP via_cnt==1". Any idea? Aimable

18 years, 9 months

[Serusers] No Audio

by Ashutosh Kumar

Hi, I am using the following ser.cfg with ser and mediaproxy, but I am getting the audio only in one direction. This does not seem a problem of NAT, as the nathelper is correclt rewriting the sdp messages, and the aor shows all clients bound with public IP. Earlier t_on_reply call in route[1] was causing an error sl_reply_error used: I'm terribly sorry, server error occured (1/SL) when one calls from one phone to another. Earlier I was getting proper bi-directional audio when I was using STUN instead of mediaproxy. Is there something wrong with my ser.cfg #-------------------------------------------------- # # $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri 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" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/acc.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/domain.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_url", "sql://ser:heslo@localhost/ser") modparam("usrloc", "db_mode", 2) modparam("uri", "db_url", "sql://ser:heslo@localhost/ser") modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") modparam("auth_radius", "service_type", 15) modparam("acc", "db_url", "sql://ser:heslo@localhost/ser") modparam("acc", "db_missed_flag", 3) modparam("acc", "db_flag", 1) modparam("acc", "log_fmt", "cdfimorstup") modparam("acc", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") modparam("acc", "service_type", 15) modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 1) modparam("acc", "log_level", 2) #modparam("acc", "radius_missed_flag", 3) modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") # -- rr params -- modparam("rr", "enable_full_lr", 1) modparam("msilo", "registrar", "sip:registrar@mydomainxyz123.com") modparam("msilo", "db_url", "sql://ser:heslo@mydomainxyz123.com/ser") modparam("auth_db", "db_url", "sql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("uri", "subscriber_table", "subscriber") modparam("uri", "uri_table", "uri") modparam("uri", "uri_user_column", "username") modparam("uri", "uri_domain_column", "domain") modparam("uri", "uri_uriuser_column", "uri_user") modparam("uri", "subscriber_user_column", "username") modparam("uri", "subscriber_domain_column", "domain") # !! Nathelper 1 #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 # !! NAthelper 1 ends # ------------------------- request routing logic ------------------- # main routing logic route{ # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- 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; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- # 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=="INVITE"){ # && client_nat_test("3")) { record_route_preset("mydomainxyz123.com:5060;nat=yes"); route(3); } else if (method!="REGISTER") {record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- # if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- lookup("aliases"); 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); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- # 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 (!method=="BYE") { force_rport(); fix_nated_contact(); }; if (uri!=myself) {route(1);break;}; if (uri==myself) { if (method=="CANCEL") {route(3); break;}; if (method=="REGISTER") {route(2); break;}; if (method=="MESSAGE") {route(4); break;}; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { if ((method=="INVITE" || method=="ACK") && t_newtran() ){ route(3);break; }; sl_send_reply("404", "Not Foundbusip1"); break; } else {setflag(1);t_relay();break;}; # if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- # t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { #end_media_session(); ; }; # sl_send_reply("404", "balleballeUser Not Found"); # sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); #if (!search("^Contact:\ +\*") && client_nat_test("7")) { # setflag(6); #fix_nated_register(); # force_rport(); #}; #if (!radius_www_authorize("")) { #if (!www_authorize("","subscriber")) { # www_challenge("","0"); # break; #}; #if (!check_to()) { # sl_send_reply("401", "Unauthorized"); # break; #}; consume_credentials(); save("location"); m_dump(); acc_rad_request("Some comment"); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- #if (client_nat_test("3")) { # setflag(7); # force_rport(); # fix_nated_contact(); #}; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); acc_db_request("404 missed call", "missed_calls"); acc_rad_request("404 missed call"); setflag(3); break; }; if (method=="CANCEL") { route(1); break; }; #if (!proxy_authorize("","subscriber")) { # proxy_challenge("","0"); # break; #} else if (!check_from()) { # sl_send_reply("403", "Use From=ID"); # break; #}; consume_credentials(); if (isflagset(6) || isflagset(7)) { # use_media_proxy(); ; }; route(1); } route[4] { if (!t_newtran()) { sl_reply_error(); break; }; if (m_store("0")) { t_reply("202", "Accepted for Later Delivery"); break; }; t_reply("503", "Service Unavailable"); break; } 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(); }; ; } Regards, Ashutosh Kumar Chetu, Inc. Ph : 1(305) 402 6724 - Witin US Ph : 91 120 5323340 - Outside US Fax:1 (305) 832 5987 For more information, please visit http://www.chetu.com

18 years, 9 months

[Serusers] Non eixsting File in mediaproxy modparam

by Ashutosh Kumar

Hi, In my ser.cfg being used with mediaproxy for NAT traversal, there are two lines reuiqred by this package. modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") But these two files do not exist (*anywhere*) on my machine. I suppose that they are a part of the mediaproxy package.Am I missing them or is the package outdated. Regards, Ashutosh Kumar Chetu, Inc. Ph : 1(305) 402 6724 - Witin US Ph : 91 120 5323340 - Outside US Fax:1 (305) 832 5987 For more information, please visit http://www.chetu.com

18 years, 9 months

[Serusers] Xlite not working with SER+SEMS

by yu wang

Hi, I've been working on conference with Xlite but it is not work. The problem is SEMS starts sending media (default wav) back to Xlite once SEMS receives the INVITE/SDP from Xlite. It seems a bug on Xlite not listening on the media port after sending the INVITE thus it replys with ICMP port unreachable. But I don't see anyone complain this on this list. Is something wrong with my configuration? Please please help me.... Thanks, Pat

18 years, 9 months

[Serusers] serctll add error

by chris

hello i tried running serctl add 1001 1001 chris_cleofe2yahoo.com but i got this error: awk: syntax error near line 1 awk: bailing out near line 1 awk: syntax error near line 1 awk: bailing out near line 1 HA1 calculation failed pls advise. thnks.

18 years, 9 months

[Serusers] ser and database

by Sebastian Kühner

Hi! Has anyone of you used a database to make routing? For example: If the caller marks 001 -> Rewritehost("xxx.xxx.xxx.111"); If the caller marks 0049 -> Rewritehost("xxx.xxx.xxx.222"); The two host names are in my postgres database... Once I used a command like this to stop calls of users that don't have credit: if (!exec_msg(' POS1=`expr index "$SIP_HF_FROM" sip: + 4`; LENGTH=`expr index "$SIP_HF_FROM" @ - $POS1`; USER=`expr substr "$SIP_HF_FROM" $POS1 $LENGTH`; QUERY="select credit from subscriber where username=$USER AND credit > 0"; RESULT=`psql -U postgres AAA -t -c "$QUERY"`; if [ -z "$RESULT" ] ; then exit 1; fi ;' )) { sl_send_reply("403", "Sorry, not enough credit to make this call."); break; }; ... but I have problems to get the result of a psql query... is there any way to do it?? Thanks!! Sebastian

18 years, 9 months

[Serusers] gw-pstn.3.07.cfg with Call-Transfer support

by Bastian Schern

Hello, I tried out the SER config "gw-pstn.3.07.cfg" from www.onsip.org. Nearly everything works fine but "Call-Transfer" is not working. Is it possible to integrate "Call-Transfer" support into the "gw-pstn.3.07.cfg"? Does anybody integrates this already? Regards Bastian debug=3 fork=yes log_stderror=no listen=192.0.2.13 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" 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/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" modparam("auth_db|permissions|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("192.0.2.13:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { 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); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(5); route(1); break; }; if (uri==myself) { if (method=="ACK") { route(6); break; } else if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; 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"); if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (method=="INVITE" && !allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (uri=~"^sip:011[0-9]*@") { # International PSTN route(4); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { # Domestic PSTN route(4); break; }; sl_send_reply("404", "User Not Found"); break; }; if (method=="CANCEL") { route(1); break; }; route(5); route(1); } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("192.0.2.245"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); 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); break; }; 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(); }; }

18 years, 9 months

[Serusers] codec question

by Innocent Evil

Scenario: IP Phone --> SIP Server --> Asterisk --> (Voice Mailbox) (G723.1, G729) -> (GSM) -> (GSM, G729) GSM coded file Is ipphone with codec g723.1 and g729 cabable to retrived voice mail from the above picture? If not, please explain a little more. Thanks a lot.

18 years, 9 months

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sr-users August 2005