sr-users October 2004

sr-users@lists.kamailio.org

189 participants
408 discussions

by md esa kamsan

Hi, I add the /ser_file/modules/mediaproxy/config/ser.cfg to the current ser.cfg and do correction for the path of modules ( .so) and try run ./ser start but there is a bad config errors. Anybody can help me? --------------------------------- Do you Yahoo!? Express yourself with Y! Messenger! Free. Download now.

20 years, 1 month

[Serusers] kphone work with ser.can NOT disconnect... ???

by dev2003＠mail.ustc.edu.cn

kphone work with ser.can NOT disconnect.kphone is no respond show KCallWidget: Starting force disconnect... then kphone is no respond . why? help me # kphone .... SipClient: Receiving message... SipClient: Received: 15:26:25.776 --------------------------------- SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.139.58;rport=15073;received=222.33.36.198 CSeq: 6935 BYE To: "24800027" <sip:t1@t1.com>;tag=e2tQMqJQRd42Gx0hev From: "wxt" <sip:t2@51t1.com>;tag=7A3E61A4 Call-ID: cc8TxjEcMBWRed(a)192.168.139.157 Server: VizufonSIP/0.2 (beta.EP.CNSTEC.Sep 15 2004) Content-Length: 0 SipCall: Incoming response SipCall: Checking for Contact and Record-Route SipTransaction: Incoming Response SipCallMember: localStatusUpdated: 200 KCallWidget: Starting force disconnect...

20 years, 1 month

[Serusers] Problems with Cisco ATA 186 + NAT and SER

by lists

Hi, I know that this problem has been discussed before, but being new to SER I am having difficulty making CISCO ATA 186 behind a CISCO 501 NAT work with my SER setup. Here is how I have it set up. +---------+ +----------------+ |cisco ata|----| cisco pix (NAT)|----+ +---------- +----------------+ | +---+ +-----------+ |SER|--+--|asterisk vm| +---+ | +-----------+ +--------------------------+ | | +-------------+ |cisco 7960 and xten phones|---------+ +--|cisco pstn gw| +--------------------------+ +-------------+ All the calls between 7960, xten, voice mail and pstn work great. When I initiate a call from ata I get audio only one way I can hear ata user but he cant hear anthing from 7960 phones or voicemail, and I am unable to initiate calls from any device to ATA. Ok its quite obvious that NAT is preventing this fom working. I tired setting up rtp proxy but that doesn't even let me register the ATA phone correctly. Can anybody help me make this work. I am including my ser.cfg file. Thanks fil ser.cfg ------------ Initial global variables debug=4 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) 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" listen=a.b.c.d mhomed=yes memlog=3 sip_warning=yes server_signature=yes #syn_branch=yes #reply_to_via=no # ------------------ module loading ---------------------------------- 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/acc.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/print.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" ##loadmodule "/usr/local/lib/ser/modules/nathelper.so" # ----------------- setting module-specific parameters --------------- modparam("tm", "fr_timer", 12) modparam("tm", "fr_inv_timer", 24) modparam("rr", "enable_full_lr", 1) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 3) modparam("usrloc", "db_mode", 2) modparam("usrloc", "timer_interval", 10) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") ##modparam("registrar", "nat_flag", 6) ##modparam("nathelper", "natping_interval", 10) ##modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT # ------------------------- request routing logic ------------------- route{ # messed up setup if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; # someonw is doing something bad if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; # record all routes if (!method=="REGISTER") record_route(); # separate the destination r-uri from the set of proxies that must be traversed loose_route(); # if the host portion of the request uri is not local, send it directly # to route processing. if (!(uri==myself)) { route(2); 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 ## if(method=="REGISTER") { ## setflag(1); # remember this is ATA ## force_rport(); ## fix_nated_contact(); ## log("LOG: In NAT clause\n"); ## fix_nated_sdp("3"); ## }; # All REGISTER attempts are processed and must always be authenticated if (method=="REGISTER") { # make sure that users don't register infinite loops if (search("^(Contact|m): .*(a)(a\.b\.c\.d\|xxxxx\.xxxxx\.com)")) { log(1, "**************** LOG: alert: someone trying to set aor==contact\n"); sl_send_reply("476", "No Server Address in Contacts Allowed" ); break; }; # digest authentcation if (!www_authorize("xxxxx", "subscriber")) { www_challenge("xxxxx", "0"); break; }; # it is an authenticated request, update Contact database now if (!save("location")) { sl_reply_error(); }; break; }; # find canonical names lookup("aliases"); #if not local domain after alias lookup forward it away if (!(uri=~"^sip:(.+@)?(a\.b\.c\.d|xxxxx\.)?xxxxx\.com)([:;\?].*)?$")) { log(1, "**************** LOG: route(5)\n"); route(5); break; }; #pstn 911, 9911, all number between 7 and 20 digits if ( (uri=~"^sip:911@.*") | (uri=~"^sip:9911@.*") | (uri=~"sip:[0-9]{7,20}@.*") ) { log(1, "**************** LOG: route(3)\n"); route(3); break; }; #voice mail if (is_user_in("Request-URI", "voicemail")) { log(1, "**************** LOG: voicemail\n"); t_on_failure("4"); setflag(4); }; if (!lookup("location")) { log(1, "**************** LOG: route(4)\n"); route(4); break; }; # check whether some inventive user has uploaded gateway # contacts to usrloc to bypass authorization logic if (uri=~"@192\.168\.0\.1|209\.208\.224\.4([;:].*)*" ) { log(1, "**************** LOG: Gateway address in UsrLoc\n"); route(3); break; }; # this flag is used with the acc module to report missed calls # to syslog. setflag(3); # do it (words to live by) append_hf("P-hint: usrloc applied\r\n"); route(1); } # ------------- process all local traffic route[1] { if (!t_relay()) { sl_reply_error(); break; }; } # ------------- process traffic leaving fikus for Internet route[2] { # outbound requests are allowed only for registered BigU users if (!(src_ip==a.b.c.d) & !(src_ip==192.168.0.15) & !(proxy_authorize("fikus", "subscriber"))) { # ACK and CANCEL have no security mechanisms so they are just # noted if (method=="ACK" | method=="BYE") { log("**************** LOG: failed outbound authentication for ACK granted\n"); } else if (method=="CANCEL") { log("**************** LOG: failed outbound authentication for CANCEL granted\n"); } else { proxy_challenge("fikus", "0"); break; }; }; # to maintain credibility of our proxy, we check From in INVITEs if (!src_ip==a.b.c.d & !src_ip==192.168.0.15 & method=="INVITE" & !check_from()) { log(1, "**************** LOG: Spoofed from attempt\n"); sl_send_reply("403", "Use From=id next time"); break; }; append_hf("P-hint: OUTBOUND ON INTERNET\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } # ------------- process traffic leaving Internet for PSTN route[3] { # all calls through the gateway must be record routed to assure # acl acceptance on the gateway record_route(); # send out emergency calls to pstn gateway immediately if ((uri=~"^sip:911@.*") | (uri=~"^sip:9911@.*")) { rewritehostport("a.b.c.e:5060"); forward(uri:host, uri:port); break; }; # seven digit numeric addresses are internal freebies sent to the pbx # without authentication if (uri=~"^sip:[0-9]{7}@(a.b.c.d|192.168.0.15|xxxxx|(xxxx\,)?\.xxxx\.com)") { rewritehostport("a.b.c.d.f:5060"); forward(uri:host, uri:port); break; }; # all numeric addresses beginning with 9 go to the pbx on the way # to the PSTN # first the caller needs to be authenticated if (uri=~"^sip:9[0-9]*@(a\.b\.c\.d|xxxxx|192\.168\.0\.15)") { if (!(src_ip==209.208.224.15 | src_ip==192.168.0.15 | method==ACK | method=="CANCEL" | method=="BYE")) { if (!proxy_authorize("xxxxx", "subscriber")) { proxy_challenge("xxxxxx","0"); break; } else if (method=="INVITE" & !check_from()) { log(1, "**************** LOG: Spoofed from attempt\n"); sl_send_reply("403", "Use From=id next time"); break; }; }; if (method=="INVITE") { # if the r-uri begins 91, does the authenticated user have # permission for long distance if (uri=~"sip:91[0-9]*@.*") { if (!is_user_in("credentials", "ld")) { sl_send_reply("403", "Local calls only"); break; }; }; }; # authenticated and authorized, now accounting is set setflag(1); }; rewritehostport("a.b.c.f:5060"); append_hf("P-hint: GATEWAY\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } # ------------- process calls for users offline route[4] { log (1, "**************** INSIDE ROUTE[4]\n"); if (!t_newtran()) { sl_reply_error(); }; if (!t_reply("404", "Not Found")) { sl_reply_error(); }; break; } # ------------- process aliased outbound traffic # inbound requests that have been aliased to a non-fikus domain # are not authenticated by fikus route[5] { append_hf("P-hint: ALIASED-OUTBOUND\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } # ------------- CC-Diversion to voicemail failure_route[4] { log (1, "**************** FAILURE_ROUTE CALLING VOICEMAIL\n"); # forward to voicemail now append_branch("sip:2000@a.b.c.d.e"); append_urihf("CC-Diversion: ", "\r\n"); append_hf("P-hint: OFFLINE-VOICEMAIL\r\n"); t_relay(); }

20 years, 1 month

[Serusers] SER -> PSTN

by Carlos Maynard

I am moving to SER from Asterisk for the radius auth and accounting. I finally made SER register and authenticate my test endpoint (Cisco ATA 186) :D i can dial to my home phone and my regular phone rings... but when i pick up i have dialtone... i'm pasting SER's output and my ser.cfg below. A codec problem? My carrier is using G729. I am moving to SER from asterisk... on asterisk i can place callls from this same ATA box trough my carrier. Does SER care what the Codec is ? Thanks in advance, Carlos Maynard Jr. ser.cfg: -------- listen=66.118.238.201 debug=9 fork=no log_stderror=no check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) fifo="/tmp/ser_fifo" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.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" modparam("usrloc", "db_mode", 0) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://ser:heslo@66.118.238.198/ser") modparam("auth_radius", "radius_config","/usr/local/etc/radiusclient/radiusclient.conf") modparam("rr", "enable_full_lr", 1) route{ log(1, "\n**********************ROUTING\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; }; record_route(); if (loose_route()) { log(1, "\n**********************LOOSE ROUTE\n"); t_relay(); break; }; if (uri==myself) { log(1, "\n**********************MYSELF\n"); if (method=="REGISTER") { log(1, "REGISTER: Authenticating user\n"); if (!radius_www_authorize("")) { log(1, "REGISTER: challenging user\n"); www_challenge("", "0"); break; }; save("location"); break; }; if (method=="INVITE") { log(1, "INVITE\n"); setflag(1); /* set for accounting (the same value as in log_flag!) */ }; if (method=="MESSAGE") { log(1, "MESSAGE\n"); setflag(1); /* set for accounting (the same value as in log_flag!) */ }; if (method=="BYE" || method=="CANCEL") { log (1, "BYE or CANCEL\n"); setflag(1); }; # IF HOST IS JUST NUMBERS SEND TO route(3) FOR E.164 PROCESSING if (uri =~ "^sip:[0-9]*@"){ log(1, "SER: matched ^sip:[0-9]*@ - just numbers - process in route(3)\n"); route(3); break; }; } } route[3] { if (uri=~"^sip:[0-9]*@") { log(1, "SER: matched ^sip:6126[1,2] - forward to GW in CANBERRA\n"); rewritehostport("38.114.15.34:5060"); t_relay_to_udp("38.114.15.34","5060"); break; }; } Console Output: ----------------- 0(13481) SIP Request: 0(13481) method: <INVITE> 0(13481) uri: <sip:12814192614@66.118.238.201;user=phone> 0(13481) version: <SIP/2.0> 0(13481) parse_headers: flags=1 0(13481) end of header reached, state=5 0(13481) parse_headers: Via found, flags=1 0(13481) parse_headers: this is the first via 0(13481) After parse_msg... 0(13481) preparing to run routing scripts... 0(13481) **********************ROUTING 0(13481) DEBUG : is_maxfwd_present: searching for max_forwards header 0(13481) parse_headers: flags=128 0(13481) end of header reached, state=9 0(13481) DEBUG: get_hdr_field: <To> [45]; uri=[sip:12814192614@66.118.238.201;user=phone] 0(13481) DEBUG: to body [<sip:12814192614@66.118.238.201;user=phone> ] 0(13481) get_hdr_field: cseq <CSeq>: <1> <INVITE> 0(13481) DEBUG: get_hdr_body : content_length=268 0(13481) found end of header 0(13481) DEBUG: is_maxfwd_present: max_forwards header not found! 0(13481) DEBUG: add_param: tag=119525115 0(13481) end of header reached, state=29 0(13481) parse_headers: flags=256 0(13481) find_first_route(): No Route headers found 0(13481) loose_route(): There is no Route HF 0(13481) check_self - checking if host==us: 14==14 && [66.118.238.201] == [66.118.238.201] 0(13481) check_self - checking if port 5060 matches port 5060 0(13481) **********************MYSELF 0(13481) INVITE 0(13481) SER: matched ^sip:[0-9]*@ - just numbers - process in route(3) 0(13481) SER: matched ^sip:6126[1,2] - forward to GW in CANBERRA 0(13481) DEBUG: t_addifnew: msg id=121 , global msg id=120 , T on entrance=0xffffffff 0(13481) parse_headers: flags=-1 0(13481) parse_headers: flags=60 0(13481) t_lookup_request: start searching: hash=30407, isACK=0 0(13481) DEBUG: proceeding to pre-RFC3261 transaction matching 0(13481) DEBUG: t_lookup_request: no transaction found 0(13481) SER: new INVITE 0(13481) parse_headers: flags=-1 0(13481) check_via_address(68.91.112.229, 192.168.1.4, 0) 0(13481) WARNING:vqm_resize: resize(0) called 0(13481) DEBUG: reply sent out. buf=0x80c5f20: SIP/2.0 1..., shmem=0x402e4040: SIP/2.0 1 0(13481) DEBUG: t_reply: finished 0(13481) check_via_address(68.91.112.229, 192.168.1.4, 0) 0(13481) DEBUG: add_to_tail_of_timer[4]: 0x402e4f30 0(13481) DEBUG: add_to_tail_of_timer[0]: 0x402e4f44 0(13481) SER: new transaction fwd'ed 0(13481) DEBUG:destroy_avp_list: destroing list (nil) 0(13481) receive_msg: cleaning up 0(13481) SIP Reply (status): 0(13481) version: <SIP/2.0> 0(13481) status: <100> 0(13481) reason: <Trying> 0(13481) parse_headers: flags=1 0(13481) Found param type 232, <branch> = <z9hG4bK7c67.dc0712e3.0>; state=16 0(13481) end of header reached, state=5 0(13481) parse_headers: Via found, flags=1 0(13481) parse_headers: this is the first via 0(13481) After parse_msg... 0(13481) forward_reply: found module tm, passing reply to it 0(13481) DEBUG: t_check: msg id=122 global id=121 T start=0xffffffff 0(13481) parse_headers: flags=17 0(13481) Found param type 234, <received> = <192.168.1.4>; state=16 0(13481) end of header reached, state=5 0(13481) parse_headers: Via found, flags=17 0(13481) parse_headers: this is the second via 0(13481) end of header reached, state=9 0(13481) DEBUG: get_hdr_field: <To> [45]; uri=[sip:12814192614@66.118.238.201;user=phone] 0(13481) DEBUG: to body [<sip:12814192614@66.118.238.201;user=phone> ] 0(13481) get_hdr_field: cseq <CSeq>: <1> <INVITE> 0(13481) parse_headers: flags=4 0(13481) DEBUG: t_reply_matching: hash 30407 label 1042378957 branch 0 0(13481) DEBUG: t_reply_matching: reply matched (T=0x402e4e08)! 0(13481) DEBUG: t_check: msg id=122 global id=122 T end=0x402e4e08 0(13481) DEBUG: t_on_reply: org. status uas=100, uac[0]=0 local=0 is_invite=1) 0(13481) ->>>>>>>>> T_code=100, new_code=100 0(13481) DEBUG: relay_reply: branch=0, save=0, relay=-1 0(13481) DEBUG: add_to_tail_of_timer[1]: 0x402e4f44 0(13481) DEBUG:destroy_avp_list: destroing list (nil) 0(13481) receive_msg: cleaning up 1(13482) DEBUG: timer routine:4,tl=0x402e4f30 next=(nil) 0(13481) SIP Reply (status): 0(13481) version: <SIP/2.0> 0(13481) status: <180> 0(13481) reason: <Ringing> 0(13481) parse_headers: flags=1 0(13481) Found param type 232, <branch> = <z9hG4bK7c67.dc0712e3.0>; state=16 0(13481) end of header reached, state=5 0(13481) parse_headers: Via found, flags=1 0(13481) parse_headers: this is the first via 0(13481) After parse_msg... 0(13481) forward_reply: found module tm, passing reply to it 0(13481) DEBUG: t_check: msg id=123 global id=122 T start=0xffffffff 0(13481) parse_headers: flags=17 0(13481) Found param type 234, <received> = <192.168.1.4>; state=16 0(13481) end of header reached, state=5 0(13481) parse_headers: Via found, flags=17 0(13481) parse_headers: this is the second via 0(13481) DEBUG: add_param: tag=SD40lb399-38 0(13481) end of header reached, state=29 0(13481) DEBUG: get_hdr_field: <To> [62]; uri=[sip:12814192614@66.118.238.201;user=phone] 0(13481) DEBUG: to body [<sip:12814192614@66.118.238.201;user=phone>] 0(13481) get_hdr_field: cseq <CSeq>: <1> <INVITE> 0(13481) parse_headers: flags=4 0(13481) DEBUG: t_reply_matching: hash 30407 label 1042378957 branch 0 0(13481) DEBUG: t_reply_matching: reply matched (T=0x402e4e08)! 0(13481) DEBUG: t_check: msg id=123 global id=123 T end=0x402e4e08 0(13481) DEBUG: t_on_reply: org. status uas=100, uac[0]=100 local=0 is_invite=1) 0(13481) ->>>>>>>>> T_code=100, new_code=180 0(13481) DEBUG: relay_reply: branch=0, save=0, relay=0 0(13481) old size: 669, new size: 606 0(13481) build_res_from_sip_res: copied size: orig:84, new: 21, rest: 585 msg= SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.4:5060;received=192.168.1.4 From: Line 0 <sip:2000@66.118.238.201;user=phone>;tag=119525115 To: <sip:12814192614@66.118.238.201;user=phone>;tag=SD40lb399-38 Call-ID: 2912392921(a)68.91.112.229 CSeq: 1 INVITE Record-Route: <sip:12814192614@66.118.238.201;ftag=119525115;lr=on> Content-Length: 176 Content-Disposition: session;handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 22631 32400 IN IP4 38.114.15.34 s=SIP Media Capabilities c=IN IP4 38.114.15.34 t=0 0 m=audio 6220 RTP/AVP 18 a=fmtp:18 annexb:no a=sendrecv a=ptime:20 0(13481) DEBUG: reply relayed. buf=0x80c67b0: SIP/2.0 1..., shmem=0x402e6840: SIP/2.0 1 0(13481) DEBUG: add_to_tail_of_timer[1]: 0x402e4f44 0(13481) DEBUG:destroy_avp_list: destroing list (nil) 0(13481) receive_msg: cleaning up 0(13481) SIP Request: 0(13481) method: <CANCEL> 0(13481) uri: <sip:12814192614@66.118.238.201;user=phone> 0(13481) version: <SIP/2.0> 0(13481) parse_headers: flags=1 0(13481) end of header reached, state=5 0(13481) parse_headers: Via found, flags=1 0(13481) parse_headers: this is the first via 0(13481) After parse_msg... 0(13481) preparing to run routing scripts... 0(13481) **********************ROUTING 0(13481) DEBUG : is_maxfwd_present: searching for max_forwards header 0(13481) parse_headers: flags=128 0(13481) end of header reached, state=9 0(13481) DEBUG: get_hdr_field: <To> [45]; uri=[sip:12814192614@66.118.238.201;user=phone] 0(13481) DEBUG: to body [<sip:12814192614@66.118.238.201;user=phone> ] 0(13481) get_hdr_field: cseq <CSeq>: <1> <CANCEL> 0(13481) DEBUG: get_hdr_body : content_length=0 0(13481) found end of header 0(13481) DEBUG: is_maxfwd_present: max_forwards header not found! 0(13481) DEBUG: add_param: tag=119525115 0(13481) end of header reached, state=29 0(13481) parse_headers: flags=256 0(13481) find_first_route(): No Route headers found 0(13481) loose_route(): There is no Route HF 0(13481) check_self - checking if host==us: 14==14 && [66.118.238.201] == [66.118.238.201] 0(13481) check_self - checking if port 5060 matches port 5060 0(13481) **********************MYSELF 0(13481) BYE or CANCEL 0(13481) SER: matched ^sip:[0-9]*@ - just numbers - process in route(3) 0(13481) SER: matched ^sip:6126[1,2] - forward to GW in CANBERRA 0(13481) DEBUG: t_addifnew: msg id=124 , global msg id=123 , T on entrance=0xffffffff 0(13481) parse_headers: flags=-1 0(13481) parse_headers: flags=60 0(13481) t_lookup_request: start searching: hash=30407, isACK=0 0(13481) DEBUG: proceeding to pre-RFC3261 transaction matching 0(13481) DEBUG: t_lookup_request: no transaction found 0(13481) DEBUG: t_lookupOriginalT: searching on hash entry 30407 0(13481) DEBUG: t_lookupOriginalT: canceled transaction found (0x402e4e08)! 0(13481) DEBUG: t_lookupOriginalT completed 0(13481) check_via_address(68.91.112.229, 192.168.1.4, 0) 0(13481) DEBUG: add_to_tail_of_timer[4]: 0x402e7040 0(13481) DEBUG: add_to_tail_of_timer[0]: 0x402e7054 0(13481) DEBUG: e2e_cancel: e2e cancel proceeding 0(13481) parse_headers: flags=-1 0(13481) check_via_address(68.91.112.229, 192.168.1.4, 0) 0(13481) WARNING:vqm_resize: resize(0) called 0(13481) DEBUG: cleanup_uacs: RETR/FR timers reset 0(13481) DEBUG: add_to_tail_of_timer[2]: 0x402e6f60 0(13481) DEBUG: reply sent out. buf=0x80c6180: SIP/2.0 2..., shmem=0x402e8420: SIP/2.0 2 0(13481) DEBUG: t_reply: finished 0(13481) DEBUG: e2e_cancel: sending 487 0(13481) parse_headers: flags=-1 0(13481) check_via_address(68.91.112.229, 192.168.1.4, 0) 0(13481) DEBUG: cleanup_uacs: RETR/FR timers reset 0(13481) DEBUG: add_to_tail_of_timer[4]: 0x402e4ebc 0(13481) DEBUG: add_to_tail_of_timer[0]: 0x402e4ed0 0(13481) DEBUG: reply sent out. buf=0x80c6550: SIP/2.0 4..., shmem=0x402e9e68: SIP/2.0 4 0(13481) DEBUG: t_reply: finished 0(13481) SER: new transaction fwd'ed 0(13481) DEBUG:destroy_avp_list: destroing list (nil) 0(13481) receive_msg: cleaning up

20 years, 1 month

[Serusers] problems about on-line users and Instant Messaging

by C Y

hello everyone: I have some problems when I configure the SIP server.I hope you can help me to solve them. The first problem: when a user login in the SIP system with X-lite or Windows Messenger 5.0,the data of field "domain"( database "ser",table "location") is NULL.It should be "voipv6.edu.cn".So I cannot find the online users through SERWEB.What should I do to solve the problem? The second problem:when two users communicate with Windows Messenger 5.0(SIP),they cannot send Instant Messaging each other.How should I configurate the SIP server to surport it? Thanks a lot! These are some informations related to my setting and problem: 1、operating system:Linux 7.0 2、SER distribution: ser-0.8.14_linux_i386.tar.gz 3、SER build: version: 0.8.14 (i386/linux) flags: STATS:Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 @(#) $Id: main.c,v 1.168.4.3 2004/06/28 15:41:21 andrei Exp $ main.c compiled on 12:28:01 Jul 27 2004 with gcc 2.95 4、SER configuration file : # # $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=yes log_stderror=no */ check_via=yes # (cmd. line: -v) dns=yes # (cmd. line: -r) rev_dns=yes # (cmd. line: -R) port=5060 #children=4 fifo="/tmp/ser_fifo" alias="voipv6.edu.cn" "210.25.130.252" "localhost" # ------------------ 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/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" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" loadmodule "/usr/local/lib/ser/modules/print.so" loadmodule "/usr/local/lib/ser/modules/textops.so" #loadmodule "/usr/local/lib/ser/modules/jabber.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/vm.so" # ----------------- setting module-specific parameters --------------- modparam("usrloc","db_url","sql://ser:heslo@localhost/ser") # -- 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", 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"); 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 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 (!www_authorize("voipv6.edu.cn", "subscriber")) { www_challenge("voipv6.edu.cn", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } _________________________________________________________________ 与联机的朋友进行交流，请使用 MSN Messenger: http://messenger.msn.com/cn

20 years, 1 month

[Serusers] SER Install problem.

by Mohamed Omar

I'm running Redhat ES on my computer and have been trying to setup SER for the last two week with very little success. The SER package that I have be using is rpm 0.8.12 for redhat 9 which I downlowad on the following locationftp://ftp.berlios.de/pub/ser/0.8.12/packages/redhat/9/ I'm just wondering if installing a rpm package for redhat 9 on redhat ES is part of the problems am having.. SER is work fine if I don't enable Digest Authentication. Please advice which package is good to install on Redhat ES. Please help... --------------------------------- Post your free ad now! Yahoo! Canada Personals

20 years, 1 month

RE: [Serusers] Ser0.8.14 + Mediaproxy

by Ricardo Martinez

Hello. I been there. :) . It takes some time to figure it out how the mediaproxy works. Anyway, i'm using - SER v 0.8.14 - Mediaproxy 1.2.0 - Python 2.3 - The ser.cfg included in /ser_directory/modules/mediaproxy/config/ is very helpfull. It's working fine to me. Good Luck Ricardo Martinez.- > -----Mensaje original----- > De: Kiko Vives [SMTP:kiko.vives@ua.es] > Enviado el: Lunes, 18 de Octubre de 2004 04:45 a.m. > Para: serusers(a)lists.iptel.org > Asunto: Re: [Serusers] Ser0.8.14 + Mediaproxy > > Hello again. > > It happens to me too. > > Can someone gives some guidelines to make it work ? I mean, version > numbers for "mediaproxy.so" and mediaproxy server, and a ser.cfg sample. > > My problem is that it compiles OK and ser run without problem but it hangs > when I try to establish a session between two UAs using the media proxy. > > Thanks ! > > Kiko > > ----- Original Message ----- > From: md esa kamsan <mailto:mesak77@yahoo.com> > To: serusers(a)lists.iptel.org <mailto:serusers@lists.iptel.org> > Sent: Monday, October 18, 2004 5:54 AM > Subject: [Serusers] Ser0.8.14 + Mediaproxy > > Hi , > > Can someone help me... where is the best guide to make it run .... > I try several times but still failed > > > > Thx > > Esa > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > <http://mail.yahoo.com> > > > _____ > > > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > <<Archivo: ATT23899.txt>>

20 years, 1 month

[Serusers] Avoiding storing passwords in mysql "subscriber" table in clear-text

by karl

Hi guys, I would appreciate if someone may help me on the subject. While still requiring users to be authenticated against user credentials (username, password, realm), on the other hand I want to avoid storing passwords in clear text in mysql "subscriber" table. Any ideas? Thank you in advanced. Best regards, Karl --------------------------------- Do you Yahoo!? vote.yahoo.com - Register online to vote today!

20 years, 1 month

Re: [Serusers] Prepaid Billing Issues (Third time)

by Nahuel Alejandro Ramos

I have already been visiting this link but all I found is commercial. I am looking for a gnu/gpl software. Thanks Mohamed. On Mon, 18 Oct 2004 17:10:59 -0400 (EDT), Mohamed Omar <amatek2004(a)yahoo.ca> wrote: > > I came across this last week, maybe it's what you want... > http://www.voip-info.org/wiki-Asterisk+Prepaid+Applications > > > > > > > Nahuel Alejandro Ramos <nahuelon(a)gmail.com> wrote: > Hi, > I looking for a prepaid biiling solution for my SER. I have already > read that I have to use B2BUA or Asterisk with some billing app. I > have a Cisco 2610 with 2 FXO so the Asterisk would not talk with the > PSTN directly. > 1- First I ask you what is better and scalable to use? > 2- How I loads or add credits to its DB? (I know that B2BUA use a > Radius, so I would use FreeRadius+MySQL) > 3- How B2BUA or Asterisk control not a user pass his credits? How > they calc the credit-time-price_perminute, and make the user hang up a > call? > 4- Do you know a free billing app, make it in PHP/MySQL for an easy > personalization? > 5- THANK YOU VERY MUCH!!! > > Nahuel Ramos. > > P.D.: I post it again by third time because the maillist was not > working right and I have not received it. > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > > > ________________________________ > Post your free ad now! Yahoo! Canada Personals > > >

20 years, 1 month

[Serusers] can't register IPPHONEs after enabling Authentication with MySQL

by Mohamed Omar

My two IP Phone were able to register to SER before I enable disgist Authentication. I capture the packets between SER and one of the phone saw that the Phone send a registration but SER replies Destination unreachable.. Below is the capture summary from Ethereal Source: 192.168.77.152 (192.168.77.152) Destination: 192.168.77.151 (192.168.77.151) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Source port: 5060 (5060) Destination port: 5060 (5060) Length: 412 Checksum: 0x3c0c (correct) Session Initiation Protocol Request-Line: REGISTER sip:sipproxy.amatek.net SIP/2.0 Method: REGISTER Resent Packet: False Message Header Via: SIP/2.0/UDP 192.168.77.152:5060;branch=z9hG4bKOphgqvT15 Max-Forwards: 70 User-Agent: iPH-D00-2G From: "777126" <sip:777126@sipproxy.amatek.net>;tag=sW0vQoefhteOBUMJ SIP Display info: "777126" SIP from address: sip:777126@sipproxy.amatek.net SIP tag: sW0vQoefhteOBUMJ To: "777126" <sip:777126@sipproxy.amatek.net> SIP Display info: "777126" SIP to address: sip:777126@sipproxy.amatek.net Call-ID: ikzXiceMcMg3NTcL(a)192.168.77.152 CSeq: 30944 REGISTER Contact: <sip:777126@192.168.77.152:5060> Expires: 30 Content-Length: 0 No. Time Source Destination Protocol Info 7 15.031286 192.168.77.151 192.168.77.152 ICMP Destination unreachable Frame 7 (474 bytes on wire, 474 bytes captured) Ethernet II, Src: 00:06:5b:6a:66:61, Dst: 00:09:45:10:22:9c Internet Protocol, Src Addr: 192.168.77.151 (192.168.77.151), Dst Addr: 192.168.77.152 (192.168.77.152) Version: 4 Header length: 20 bytes Differentiated Services Field: 0xc0 (DSCP 0x30: Class Selector 6; ECN: 0x00) Total Length: 460 Identification: 0xaef6 (44790) Flags: 0x00 Fragment offset: 0 Time to live: 64 Protocol: ICMP (0x01) Header checksum: 0xacfa (correct) Source: 192.168.77.151 (192.168.77.151) Destination: 192.168.77.152 (192.168.77.152) Internet Control Message Protocol Type: 3 (Destination unreachable) Code: 3 (Port unreachable) Checksum: 0x1b2b (correct) Internet Protocol, Src Addr: 192.168.77.152 (192.168.77.152), Dst Addr: 192.168.77.151 (192.168.77.151) Version: 4 Header length: 20 bytes Differentiated Services Field: 0x00 (DSCP 0x00: Default; ECN: 0x00) Total Length: 432 Identification: 0x00e6 (230) Flags: 0x00 Fragment offset: 0 Time to live: 128 Protocol: UDP (0x11) Header checksum: 0x1bd7 (correct) Source: 192.168.77.152 (192.168.77.152) Destination: 192.168.77.151 (192.168.77.151) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Source port: 5060 (5060) Destination port: 5060 (5060) Length: 412 Checksum: 0x3c0c (correct) Session Initiation Protocol Request-Line: REGISTER sip:sipproxy.amatek.net SIP/2.0 Method: REGISTER Resent Packet: False Message Header Via: SIP/2.0/UDP 192.168.77.152:5060;branch=z9hG4bKOphgqvT15 Max-Forwards: 70 User-Agent: iPH-D00-2G From: "777126" <sip:777126@sipproxy.amatek.net>;tag=sW0vQoefhteOBUMJ SIP Display info: "777126" SIP from address: sip:777126@sipproxy.amatek.net SIP tag: sW0vQoefhteOBUMJ To: "777126" <sip:777126@sipproxy.amatek.net> SIP Display info: "777126" SIP to address: sip:777126@sipproxy.amatek.net Call-ID: ikzXiceMcMg3NTcL(a)192.168.77.152 CSeq: 30944 REGISTER Contact: <sip:777126@192.168.77.152:5060> Expires: 30 Content-Length: 0 --------------------------------- Post your free ad now! Yahoo! Canada Personals

20 years, 1 month

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sr-users October 2004