- sr-users - kamailio.org

[OpenSER-Users] libwwwxml notsupporting Openser
by A Chaturvedi 17 Aug '07

17 Aug '07

Hi, I was trying to use OpenSer and with its previous versions it worked fine but with new version that has xml support(as said)it fails giving this error: openser 0(15681) ERROR: load_module: could not open module </lib/openser/modules/mi_xmlrpc.so>: /usr/lib/libwwwxml.so.0: undefined symbol: WWW_TraceFlag 0(15681) parse error (60,13-14): failed to load module 0(15681) set_mod_param_regex: No module matching mi_xmlrpc found | 0(15681) parse error (78,22-23): Can't set module parameter ERROR: bad config file (2 errors) 0(15681) INFO:mi_fifo:mi_destroy:memory for the child's mi_fifo_pid was not allocated -> nothing to destroy I have installed w3c-libwww pakage as well as its devel package still same error occurs.. Xmlrpc package is also properly installed. Entire things get stuck at:/usr/lib/libwwwxml.so.0: undefined symbol: WWW_TraceFlag Hope someone can help.. Ashish Chaturvedi Mailto: a.chaturvedi(a)tcs.com =====-----=====-----===== Notice: The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail or telephone and immediately and permanently delete the message and any attachments. Thank you

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[OpenSER-Users] Notify behaviour?
by Анонім Анонім 17 Aug '07

17 Aug '07

I use openser presence module for presence. And I have some troubles with it. When subscribing to presence of offline user “handle_subscribe” function send “202OK” response, and “NORIFY” without content-type header field. Is this normal behaviour, or I missed something in openser configuration? ___________________________________________________ http://www.bigmir.net - поиск по украинским сайтам

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[OpenSER-Users] rtpproxy
by Kent Poppenhouse 16 Aug '07

16 Aug '07

I have a question about the nathelper module. specifically the rtpproxy_disable_tout, rtpproxy_retr, and rtpproxy_tout parameters. I have openser running on one machine and rtpproxy on another. Everything works great if the rtpproxy is up and running before starting openser. However, if rtpproxy is not running first, openser never connects to it. I set the following parameters in my openser config file: # -- nat params -- modparam("nathelper", "rtpproxy_sock", "udp:10.100.103.1:8899") modparam("nathelper", "rtpproxy_disable_tout", 30) modparam("nathelper", "rtpproxy_tout", 2) modparam("nathelper", "rtpproxy_retr", 10) based on this I would think openser would try to connect to the rtpproxy 10 times with a 2 sec timeout. If after 10 retries no connection is made, it would wait 30 s before attempting to connect again. Is this correct? If so, what am I missing since it doesn't do this. I'm using openser version 1.2.0. Thanks.

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[OpenSER-Users] OpenSER v1.2.2 released
by Daniel-Constantin Mierla 16 Aug '07

16 Aug '07

Hello, version 1.2.2 of OpenSER has been released. This is a minor release in the branch of 1.2 release series, containing only fixes to bugs found since the previous release 1.2.1. See all changes at: http://www.openser.org/pub/openser/1.2.2/ChangeLog The configuration script used with any 1.2.x version is compatible. There was a change in the structure of table "active_watchers" for handling local and remote CSeq numbers -- you have to recreate this table if you are using presence module (instructions are provided here: http://openser.org/dokuwiki/doku.php/install:1.2.1-to-1.2.2). Source tarball can be downloaded from: http://www.openser.org/pub/openser/latest/src/ Or from sourceforge download page: http://sourceforge.net/project/showfiles.php?group_id=139143 Download page providing several options is available at: http://www.openser.org/mos/view/Download/ The documentation for modules is posted at: http://openser.org/docs/modules/1.2.x/ Main documentation page is available at: http://www.openser.org/mos/view/OpenSER-Documentation-Repository/ If someone wants to contribute with packages for different distros, please send the link from where can be downloaded to team(a)openser.org. We would like to thank to all developers and contributors for their work with coding and testing for this new release. Cheers, Daniel

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[Serusers] Presence annoyance.
by SIP 16 Aug '07

16 Aug '07

There is now a Grandstream GXV3000 in my network (neat little videophones, I must say). However, when someone sends a presence event through my server, the grandstream responds with a SIP packet and additional XML info: U 99.101.102.103:5060 -> 172.17.240.240:54330 NOTIFY sip:1101XXXXXXX@172.17.240.240:54330 SIP/2.0. Record-Route: <sip:99.101.102.103;ftag=f0a1719429e37564;lr=on>. Via: SIP/2.0/UDP 99.101.102.103;branch=z9hG4bK4204.13bfa134.0. Via: SIP/2.0/UDP 66.69.71.90:34202;branch=z9hG4bK04c43fa29c41b774. From: <sip:1101ZZZZZZZ@proxy.ourdomain.com>;tag=f0a1719429e37564. To: "Jessa"<sip:1101XXXXXXX@proxy.ourdomain.com>;tag=1b55e40b. Contact: <sip:1101ZZZZZZZ@66.69.71.90:34202>. Call-ID: ZjczZGMzOWEzMTk0NjhkMzRjMjI3ODZkNzIwMGIxZDA.. CSeq: 1843 NOTIFY. User-Agent: Grandstream GXV3000 (HW 0101, Ch:0) 1.0.1.7. Max-Forwards: 16. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK. Event: presence. Subscription-State: active. Content-Type: application/xpidf+xml. Content-Length: 342. P-hint: LR. . <?xml version="1.0"?>. <!DOCTYPE presence. PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">. <presence>. <presentity uri=""Jessa"<sip:1101XXXXXXX@proxy.ourdomain.com>;tag=1b55e40b;method=SUBSCRIBE" />. <atom atomid="1000">. <address uri="<sip:1101ZZZZZZZ@66.69.71.90:34202>">. <status status="open" />. </address>. </atom>. </presence> The XML info is causing errors in the logs (which I've been ignoring for now, but that's not really an optimal solution). Aug 15 15:16:12 proxy ser[19620]: ERROR: parse_to : unexpected char [>] in statu s 4: <<<>> . Aug 15 15:16:12 proxy ser[19620]: ERROR: get_hdr_field: bad to header Aug 15 15:16:12 proxy ser[19620]: ERROR:maxfwd:is_maxfwd_present : parsing MAX_F ORWARD header failed! Aug 15 15:16:12 proxy ser[19620]: ERROR: parse_uri: uri too short: <> (0) Aug 15 15:16:12 proxy ser[19620]: is_user_in(): Error while parsing URI Aug 15 15:16:12 proxy ser[19620]: ERROR: parse_to : unexpected char [>] in statu s 4: <<<>> . Aug 15 15:16:12 proxy ser[19620]: ERROR: get_hdr_field: bad to header Aug 15 15:16:12 proxy ser[19620]: find_first_route: Error while parsing headers Aug 15 15:16:12 proxy ser[19620]: ERROR: parse_uri: uri too short: <> (0) Aug 15 15:16:12 proxy ser[19620]: ERROR: parse_sip_msg_uri: bad uri <> Aug 15 15:16:12 proxy ser[19620]: WARNING: do_action:error in expression Aug 15 15:16:12 proxy ser[19620]: ERROR: parse_to : unexpected char [>] in statu s 4: <<<>> . Aug 15 15:16:12 proxy ser[19620]: ERROR: get_hdr_field: bad to header Aug 15 15:16:12 proxy ser[19620]: ERROR: parse_to : unexpected char [>] in statu s 4: <<<>> . Aug 15 15:16:12 proxy ser[19620]: ERROR: get_hdr_field: bad to header Aug 15 15:16:12 proxy ser[19620]: find_credentials(): Error while parsing header s Aug 15 15:16:12 proxy ser[19620]: pre_auth(): Error while looking for credential s Aug 15 15:16:12 proxy ser[19620]: ERROR: parse_to : unexpected char [>] in statu s 4: <<<>> . Aug 15 15:16:12 proxy ser[19620]: ERROR: get_hdr_field: bad to header Aug 15 15:16:12 proxy ser[19620]: ERROR: parse_to : unexpected char [>] in statu s 4: <<<>> . Aug 15 15:16:12 proxy ser[19620]: ERROR: get_hdr_field: bad to header Aug 15 15:16:12 proxy ser[19620]: ERROR: build_res_buf_from_sip_req: alas, parse _headers failed Aug 15 15:16:12 proxy ser[19620]: pre_auth(): Error while sending 400 reply Other than forwarding presence information to a presence-capable server that understands the XML, is there anything I can do on the SER side to ignore the XML stuff so I don't get a screen full of error messages every time a presence event comes through? N.

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[Serusers] presence server 2 presence server
by cherami＠gmx.de 16 Aug '07

16 Aug '07

Hi, I am student and new to the Presence Model und Protocols. I have in my student project the following scenario: company A wants to develop a presence service PA. The presence service PA should get the data from the presence server PB of an other Company B over the Internet. The Presence information of PA should be made available in the Company A for different applications. my question: can PA poll or subscribe all changed/updated Presence Information from PB regularly. Or it should request the Presence Information of each Presentity individually? which Presence server can implement the presence service PA. What should I do if PB uses another Protokoll? Thank you and Best Regards Koder -- Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kanns mit allen: http://www.gmx.net/de/go/multimessenger

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[Serusers] ser asterisk pstn
by ronald＠stanford.edu 16 Aug '07

16 Aug '07

hello all, I am stuck and need your help for +xlite +ser +asterisk +pstn calls. I dial 914082743304 and get "caller Not Found". I am able to authenticate and complete sip -> sip calls, but cannot complete PSTN calls I did monitor asterisk and no reroute was offered. please help! permissions.allow: ^sip:91[0-9]{10}@bigu\.edu$ ^sip:9011[0-9]*@bigu\.edu$ ^sip:9[0-9]{10}@bigu\.edu$ ngrep -vvvv proto udp port 5060 produces the following response: 171.66.176.123:47729 -> 192.168.11.243:5060 INVITE sip:914082743304@bigu.edu SIP/2.0..Via: SIP/2.0/UDP 192.168.1.100:47729;br anch=z9hG4bK-d87543-4f5eb56fc40f1f43-1--d87543-;rport..Max-Forwards: 70..Contact: <si p:cell@192.168.1.100:47729>..To: "914082743304"<sip:914082743304@bigu.edu>..From: "SER EXpress"<sip:cell@bigu.edu>;tag=b9283f27..Call-ID: Y2JkY2ZmMDQyMjQ3YTljYTg5 NTdlOGFiNGUyZWVlY2Q...CSeq: 1 INVITE..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER , NOTIFY, MESSAGE, SUBSCRIBE, INFO..Content-Type: application/sdp..User-Agent: eyeBea m release 1011s stamp 41121..Content-Length: 421....v=0..o=- 1 2 IN IP4 192.168.1.100 ..s=CounterPath eyeBeam 1.5..c=IN IP4 192.168.1.100..t=0 0..m=audio 32942 RTP/AVP 107 100 106 6 0 105 8 18 3 5 101..a=fmtp:18 annexb=yes..a=fmtp:101 0-15..a=rtpmap:107 BV 32/16000..a=rtpmap:100 SPEEX/16000..a=rtpmap:106 SPEEX-FEC/16000..a=rtpmap:105 SPEEX- FEC/8000..a=rtpmap:18 G729/8000..a=rtpmap:101 telephone-event/8000..a=sendrecv..a=x-r tp-session-id:AEB31A44A04C4B32BD3D72349E4FBB80.. # U 192.168.11.243:5060 -> 171.66.176.123:47729 SIP/2.0 404 caller Not Found..Via: SIP/2.0/UDP 192.168.1.100:47729;branch=z9hG4bK-d875 43-4f5eb56fc40f1f43-1--d87543-;rport=47729;received=171.66.176.123..To: "914082743304 "<sip:914082743304@bigu.edu>;tag=21f9bfca23409407a263e4a7e40abe9d.f2a8..From: "SE R EXpress"<sip:cell@bigu.edu>;tag=b9283f27..Call-ID: Y2JkY2ZmMDQyMjQ3YTljYTg5NTdl OGFiNGUyZWVlY2Q...CSeq: 1 INVITE..Server: Sip EXpress router (0.9.4 (i386/linux))..Co ntent-Length: 0..Warning: 392 192.168.11.243:5060 "Noisy feedback tells: pid=15307 re q_src_ip=171.66.176.123 req_src_port=47729 in_uri=sip:914082743304@bigu.edu out_u ri=sip:914082743304@bigu.edu via_cnt==1".... debug=3 fork=yes log_stderror=no listen=1xx.x4.11.xx3 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 alias=sip2.bigu.edu dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser[1]" 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/domain.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/permissions.so" modparam("auth_db|permissions|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser[2]") 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") #added these timers for voicemail forwarding modparam("tm", "fr_inv_timer", 25) modparam("tm", "noisy_ctimer", 1) 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("1xx.x4.11.xx3:5060;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 ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); break; }; if (method=="INVITE") { if (!allow_trusted()) { if (!proxy_authorize("sip2.bigu.edu","subscriber")) { proxy_challenge("sip2.bigu.edu","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; if (client_nat_test("3")||search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { sl_send_reply("403", "Forbidden"); }; break; }; if (method=="ACK") { route(1); break; } else if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (!is_uri_host_local()) { route(4); 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("sip2.bigu.edu","subscriber")) { www_challenge("sip2.bigu.edu","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (!allow_trusted()) { if (!proxy_authorize("sip2.bigu.edu","subscriber")) { proxy_challenge("sip2.bigu.edu","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; if (uri=~"^sip:1[0-9]{10}@bigu.edu") { strip(1); }; lookup("aliases"); if (!is_uri_host_local()) { route(4); route(1); break; }; if (uri=~"^sip:011[0-9]*@bigu.edu") { # International PSTN route(4); route(5); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@bigu.edu") { # Domestic PSTN route(4); route(5); break; }; sl_send_reply("404", "caller Not Found"); break; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { if (!isflagset(8)) { setflag(8); use_media_proxy(); }; }; } route[5] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("1xx.x4.8.xx8"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); 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(); }; } Links: ------ [1] mysql://ser:heslo@localhost/ser [2] mysql://ser:heslo@localhost/ser

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[OpenSER-Users] CallManager Express + Exchange 2007
by Brian D 16 Aug '07

16 Aug '07

Would anyone be willing to share their configuration of where they have integrated these two using OpenSER? Thanks - b

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[Serusers] Out of Office
by chayn＠imhocorp.com 16 Aug '07

16 Aug '07

Hello, I am out of the office and will be unable to respond to messages until Thursday, August 16. Claude

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[Serusers] ser to sip provider
by julia heller 16 Aug '07

16 Aug '07

Hi, i attempt to connect ser to a sip provider with user + password authentication. if i get it right, i have to load the uac modul and insert following line: modparam("uac","credential","user:sip-provider-ip:password"), but i can't find the uac modul .. i hope anybody can help me .. another way i tried is this one: route[x] { rewriteuri("username@sip-provider-ip"); rewriteuserpass("username:password"); rewritehost("sip-provider-ip"); forward("username:password@sip-provider-ip"); } but it worked either. it comes always up with: get_connection(): Connection not found in the pool .. kind regards, julia heller -- Der GMX SmartSurfer hilft bis zu 70% Ihrer Onlinekosten zu sparen! Ideal für Modem und ISDN: http://www.gmx.net/de/go/smartsurfer

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[OpenSER-Users] question about DIALOG module
by Victor Gamov 15 Aug '07

15 Aug '07

Hi All As I understand dialog may be terminated by DIALOG module, e.g. due session timeout. In this situation module sends BYE timeout. When RTPproxy used does it receive signal to close RTP session whel local BYE generated? One more question -- is it possible to terminate session when RTPproxy does not get RTP packets for some predefined time? may be RTPproxy must be extended? Thanks! -- CU, Victor Gamov

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[Serusers] Regarding SNMP and Statistics in SER
by Parminder.Nayyar＠relianceada.com 15 Aug '07

15 Aug '07

Hi friends I have been on the look out for someting related to performance counters or statistics in SER. If i browse through the code, I find, that there is #ifdef STAT macro defined, which is used if STATs are enabled while compilation. Now, when i try to do that, it looks for a SNMP module related files... which it doesnt find... On searching on internet i find a page saying SNMP is dead... Can any one help me in finding out how to enable statistics for SER? It is extremely urgent for me... please help .. :(( Thanks Parminder Nayyar The information contained in this e-mail message is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you should return it to the sender immediately. Please note that while we scan all e-mails for viruses we cannot guarantee that any e-mail is virus-free and accept no liability for any damage caused by any virus transmitted by this email.

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[Serusers] routing between ser and asterisk
by Steffen Waldmann 15 Aug '07

15 Aug '07

hi there, i have a question about routing between SER and asterisk: I run SER and * on the same machine, so i configured * to listen on port 5070 (sip.conf: bindport 5070). Now i want to route all numbers that are not found in SER to *, therefore i wrote to ser.cfg in the main route section: if (!lookup("location")) { #sl_send_reply("404", "nicht gefunden"); rewritehostport("192.168.2.183:5070"); route(1); break; }; route(1) is the following: route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (!t_relay()) { sl_reply_error(); }; } i have two SER-users: 1000 and 1001 i have one * user: 1002 when i dial 1002 i get immediately 'User not found' does anyone have a hint?

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[Serusers] Timer between Trying and 18x
by Fernando Schmitt 15 Aug '07

15 Aug '07

Guys, I modified some code in order do create a timer to control the time between Trying and 18x. Do you think it's usefull? In case you think, I would like to contribute. Fernando Schmitt

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[Serusers] SER Proxy with Redirect
by Fernando Schmitt 15 Aug '07

15 Aug '07

Juha, I'm new into C programming. Could you kindly let me know where this code should be inserted? yes, i implemented a function called move that turned 302 to invite, but that code relied on a patch to tm module that gave me access to contact header of reply in reply route. i found the function and it is below. someone who needs this functionality has to check if same kind of access to the contact header of the reply is still possible or if the access method is now different and modify the code accordingly. also, all contacts would need to be taken from 302, not just the first. contact_re was compiled during module initialization: regcomp(&contact_re, "^Contact:(.*)$", REG_EXTENDED|REG_NEWLINE); needless to say, i'm not proud of this code, but it gives an idea on how it can be done. -- juha /* * Appends a new branch to request using the first Contact URI of the (302) * reply (if any) as the Request URI. Returns 1 if appending succeeded and -1 * if it failed. */ int move(struct sip_msg* _m, char* _s1, char* _s2) { struct sip_msg* reply; regmatch_t pmatch[2]; struct hdr_field hf; contact_t* first; reply = _m->final_reply; if (!reply) { LOG(L_ERR, "move(): No reply found\n"); return -1; } LOG(L_ERR, "move(): unparsed part of reply: %s\n", reply->unparsed); if ((regexec(&contact_re, reply->unparsed, 2, &(pmatch[0]), 0) != 0) || (pmatch[1].rm_so == -1)) { LOG(L_ERR, "move(): No Contact header found\n"); return -1; } hf.type = HDR_CONTACT; hf.name.len = 0; hf.body.len = pmatch[1].rm_eo - pmatch[1].rm_so - 1; hf.body.s = &(reply->unparsed[pmatch[1].rm_so]); hf.len = hf.body.len + 2; hf.parsed = NULL; hf.next = NULL; LOG(L_ERR, "hf.body: '%.*s'\n", hf.body.len, hf.body.s); if (parse_contact(&hf) < 0) { LOG(L_ERR, "move(): Error while parsing Contact\n"); goto failure; } if (((contact_body_t*)hf.parsed)->star == 1) { LOG(L_ERR, "move(): Contact is *\n"); goto failure; } first = ((contact_body_t*)hf.parsed)->contacts; if (first) { if (append_branch(_m, first->uri.s, first->uri.len) == 1) { goto success; } else { LOG(L_ERR, "move(): Appending branch failed\n"); } } else { LOG(L_ERR, "move(): No contacts in Contact header\n"); } failure: if (hf.parsed) { free_contact((contact_body_t**)(&(hf.parsed))); } return -1; success: free_contact((contact_body_t**)(&(hf.parsed))); return 1; } Fernando Schmitt

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[OpenSER-Users] tm.so module
by Brian D 15 Aug '07

15 Aug '07

I am unable to call the t_relay_to_tcp() or t_relay_to_udp() functions even though the tm.so module is being loaded. Any thoughts? Thanks, - b

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[Serusers] Out of Office
by chayn＠imhocorp.com 15 Aug '07

15 Aug '07

Hello, I am out of the office and will be unable to respond to messages until Thursday, August 16. Claude

1 0

[OpenSER-Users] Not insert Via header on forwarding/replicate/t_relay etc?
by Pat wang 15 Aug '07

15 Aug '07

Hi all, Is there anyway that I can use forward, t_replicate, or t_relay command without inserting a Via header? What I am trying to do is to use OpenSER as a load balancer but just filtering incoming messages and distribute to real server but not adding an extra Via header. Thanks for your help, Patrick _________________________________________________________________ Messenger Café open for fun 24/7. Hot games, cool activities served daily. Visit now. http://cafemessenger.com?ocid=TXT_TAGHM_AugHMtagline

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[OpenSER-Users] To configure TLS support
by Vikas CG. 15 Aug '07

15 Aug '07

Hi all, Can any help me out in running openser server with TLS support? In openser conf file they have commented for TLS and given comment that we must uncomment it to run TLS support but unfortunately while I compile it's giving configuration error, have I want to do some more settings. Thanks, Vikas C.G. -----Original Message----- From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Marc LEURENT Sent: Thursday, August 09, 2007 7:19 PM To: flavio Cc: users(a)openser.org Subject: Re: [OpenSER-Users] Re: How can it be simpler and working? CDRTool YEAHHHHHH! My problem was with the destination database. I didn't understand what was the destination ID, I thought it was just a random one! I was probably too tired... Hopefully, tomorrow; it's summer holidays I owe you a beer!! Are you going to Rome next September for VoIP conference? ShowPrice From=sip:101@sd-7501.dedibox.fr To=sip:001305465768 Duration=599876 199.9587 App: audio Destination: 1305 (TEST INTERNATIONAL) Customer: default Connect fee: 0.0000 -- Span: 1 Duration: 11931 s StartTime: 2007-08-09 15:41:09 (Europe/Paris) ProfileId: 441 for weekday RateId: 441 for 8-19h Rate: 0.0200 / 60 s Price: 3.9770 -- Span: 2 Duration: 29931 s StartTime: 2007-08-09 15:41:09 (Europe/Paris) ProfileId: 441 for weekday RateId: 442 for 19-24h Rate: 0.0200 / 60 s Price: 9.9770 -- Span: 3 Duration: 58731 s StartTime: 2007-08-09 15:41:09 (Europe/Paris) ProfileId: 441 for weekday RateId: 442 for 0-8h Rate: 0.0200 / 60 s Price: 19.5770 Yes: "reportMissingRates"=> 1 flavio a écrit : > Marc, > > Is the missing rates parameter of $CDRTool['rating'] equal to 1? > > rates.csv > Ops,Gateway,Domain,Subscriber,Rate Id,Destination,Price,App,Connect > 2,,192.168.1.160,,442,1305,200,audio, > 2,,192.168.1.160,,441,1305,200,audio, Ops,Gateway,Domain,Subscriber,Rate Id,Destination,Price,App,Connect 2,,sd-7501.dedibox.fr,,442,1305,200,audio, 2,,sd-7501.dedibox.fr,,441,1305,200,audio, > > > profiles.csv > Ops,Gateway,Domain,Subscriber,Profile Id,Rate Id1,00-H1,Rate > Id2,H1-H2,Rate Id3,H2-H3,Rate Id4,H3-24 > 2,,192.168.1.160,,441,442,8,441,19,442,24,,0 > 2,,192.168.1.160,,442,442,24,442,19,441,24,,0 Ops,Gateway,Domain,Subscriber,Profile Id,Rate Id1,00-H1,Rate Id2,H1-H2,Rate Id3,H2-H3,Rate Id4,H3-24 2,,sd-7501.dedibox.fr,,442,442,24,442,19,442,24,,0 2,,sd-7501.dedibox.fr,,441,442,8,441,19,442,24,,0 > > > customer.csv > > Ops,Gateway,Domain,Subscriber,WeekDay,Fallback,WeekEnd,Fallback,Timezone,Incr,Minim > > 2,,,,441,,442,,Europe/Amsterdam,0,0 Ops,Gateway,Domain,Subscriber,WeekDay,Fallback,WeekEnd,Fallback,Timezone,Incr,Minim 2,,,,441,,442,,Europe/Paris,0,0 > > > I have made all the tests dialing 001305...... (Miami). (I have used > 192.168.1.160 as the domain, change to yourdomain). It normalizes to > 1305 striping two zeros as specified on cdrlib.phtml. Check if it is > striping the digits to match the desired destination. > > class CDRS { > > var $CDR_class = 'CDR'; > var $intAccessCode = '00'; > var $natAccessCode = '0'; > class CDRS { var $CDR_class = 'CDR'; var $intAccessCode = '00'; var $natAccessCode = '0'; var $RatingEngineSocket = false; var $maxrowsperpage = 15; var $status = array(); var $normalizedField = 'Normalized'; var $DestinationIdField = 'DestinationId'; var $BillingIdField = 'UserName'; var $sipTraceDataSource = 'sip_trace'; var $defaults = array(); var $whereUnnormalized = ''; ...................... ...................... > Cheers, > > Flavio > ----- Original Message ----- From: "Marc LEURENT" <lftsy(a)free.fr> > To: "flavio" <flavio(a)asteriskguide.com> > Sent: Thursday, August 09, 2007 9:41 AM > Subject: Re: [OpenSER-Users] Re: How can it be simpler and working? CDRTool > > >> Hello flavio, >> I would be very grateful if you can send me just the .csv of your >> tables about your internal calls, the simpliest one for you that is >> working... >> Best Regards! >> >> Marc, desperate >> >> Marc LEURENT a écrit : >>> I've got a running tail -f /var/lig/messages and I haven't seen a >>> single error... >>> It's just that it calculates 0.0000 for the price >>> >>> I have already added the $CDRTool['rating'] parameter.. >>> >>> Id don't have the same message: >>> Aug 9 14:00:02 sd-7501 CDRTool[18946]: ShowPrice Id=3 >>> Timestamp=1186660750 Duration=11 inputTraffic=0 outputTraffic=0 >>> From=sip:106@sd-7501.dedibox.fr >>> To=6713(a)sd-7501.dedibox.fr Gateway= Application=audio >>> Aug 9 14:00:02 sd-7501 CDRTool[19627]: Normalization done in 0 s, >>> memory usage: 9.90 MB >>> Aug 9 14:00:02 sd-7501 CDRTool[19627]: 1 CDRs, 1 normalized in 0 s @ >>> 0 cps >>> >>> But there is no "ConnectFee=0.0000 Span=1 Duration=57 De >>> stId=1305 domain=192.168.1.160 Profile=441 Period=weekday Rate=441 >>> Interval=8 >>> -19 Cost=0.0200/10 Price=0.1140 " after >>> >>> Do I have to activate something else?? >>> >>> >>> flavio a écrit : >>>> Marc, >>>> >>>> I'm using 5.1.5 and it is working. Check the /var/log/syslog for errors >>>> in the normalization process >>>> >>>> Something like: >>>> >>>> Aug 6 07:18:55 debian CDRTool[3973]: Error: Cannot find rates for >>>> callid=, d >>>> omain=192.168.1.160, destination=1305, profile=441, app=audio >>>> >>>> The correct in the tests I have made is: >>>> >>>> Aug 6 07:47:44 debian CDRTool[3973]: ShowPrice Id=88 >>>> Timestamp=1186400781 Du >>>> ration=57 inputTraffic=0 outputTraffic=0 From=1000(a)192.168.1.160 >>>> To=001305345 >>>> 6789(a)192.168.1.160 Gateway=192.168.1.118 Application=audio >>>> Aug 6 07:47:44 debian CDRTool[3973]: ConnectFee=0.0000 Span=1 >>>> Duration=57 De >>>> stId=1305 domain=192.168.1.160 Profile=441 Period=weekday Rate=441 >>>> Interval=8 >>>> -19 Cost=0.0200/10 Price=0.1140 >>>> >>>> please activate the following parameter in the global.inc file >>>> (reportMissingRates). >>>> >>>> $CDRTool['rating']=array("priceDenominator" => >>>> 10000,"priceDecimalDigits" => 4,"minimumDurationCharged"=> >>>> 0,"durationPeriodRated"=> 60,"trafficSizeRated"=> >>>> 1024,"reportMissingRates"=> 1,"minimumDuration"=> 0); >>>> >>>> I hope it helps, it took me a long time before to start working. >>>> >>>> Cheers >>>> >>>> Flavio >>>> >>>> >>>> ----- Original Message ----- From: "Marc LEURENT" <lftsy(a)free.fr> >>>> To: "flavio" <flavio(a)asteriskguide.com> >>>> Cc: <users(a)openser.org> >>>> Sent: Thursday, August 09, 2007 9:01 AM >>>> Subject: Re: [OpenSER-Users] Re: How can it be simpler and working? >>>> CDRTool >>>> >>>> >>>>> Yes, I've added all rights to my account... >>>>> and price is NULL in the database until a Normalize is done! All >>>>> prices are set to 0.0000 after... >>>>> >>>>> And after a >>>>> Normalize datasource ser_radius, database DB_radius, table >>>>> radacct200708 >>>>> Aug 9 14:00:02 sd-7501 CDRTool[18946]: ShowPrice Id=3 >>>>> Timestamp=1186660750 Duration=11 inputTraffic=0 outputTraffic=0 >>>>> From=sip:106@sd-7501.dedibox.fr >>>>> To=6713(a)sd-7501.dedibox.fr Gateway= Application=audio >>>>> >>>>> >>>>> >>>>> >>>>> flavio a écrit : >>>>>> Marc, >>>>>> >>>>>> Have you checked your user rights in the CDRTool (is show price item >>>>>> highlighted?). My CDRtool shown x.xxx in the price and that's was >>>>>> the >>>>>> cause. >>>>>> >>>>>> Flavio >>>>>> >>>>>> ----- Original Message ----- From: "Marc LEURENT" <lftsy(a)free.fr> >>>>>> To: "Dan-Cristian Bogos" <dan.bogos(a)gmail.com> >>>>>> Cc: <users(a)openser.org> >>>>>> Sent: Thursday, August 09, 2007 8:01 AM >>>>>> Subject: [OpenSER-Users] Re: How can it be simpler and working? >>>>>> CDRTool >>>>>> >>>>>> >>>>>> I'm starting to fell sorry to bother you like that... >>>>>> I have changed the database like you said, but it is still not >>>>>> working... >>>>>> >>>>>> ShowPrice From=sip:101@sd-7501.dedibox.fr To=0033677832974 >>>>>> Duration=5923 >>>>>> 0.0000 >>>>>> >>>>>> Should I try to downgrade to version 5.1.3 like yours? >>>>>> >>>>>> ShowMemory >>>>>> 1 destinations >>>>>> 1 customers >>>>>> 1 profiles >>>>>> 1 rates >>>>>> 0 holidays >>>>>> 2 prepaid accounts >>>>>> 1 sessions >>>>>> 1 sessions last minute >>>>>> 1 sessions last hour >>>>>> 1 sessions last day >>>>>> >>>>>> >>>>>> but in the database, the prepaid account is credited by 10.00 >>>>>> ShowPrepaidAccounts Filter=101(a)sd-7501.dedibox.fr >>>>>> Account=101(a)sd-7501.dedibox.fr Balance=10040.0000 >>>>>> >>>>>> ShowCustomers >>>>>> default >>>>>> >>>>>> >>>>>> ShowAccounts >>>>>> Error: Invalid request >>>>>> >>>>>> MaxSessionTime From=sip:101@sd-7501.dedibox.fr >>>>>> To=sip:106@sd-7501.dedibox.fr Duration=7200 Lock=1 >>>>>> none >>>>>> >>>>>> DebitBalance From=sip:101@sd-7501.dedibox.fr >>>>>> To=sip:106@sd-7501.dedibox.fr Duration=59 >>>>>> Ok >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Dan-Cristian Bogos a écrit : >>>>>>> Ok, let's take it in your way. I would rather modify your data into: >>>>>>> >>>>>>> Destinations: >>>>>>> Ops,Gateway,Domain,Subscriber,Destination Id,Description >>>>>>> 2,,,,131,TEST INTERNATIONAL >>>>>>> >>>>>>> Customers: >>>>>>> Ops,Gateway,Domain,Subscriber,WeekDay,Fallback,WeekEnd,Fallback,Timezone,Incr,Minim >>>>>>> >>>>>>> >>>>>>> >>>>>>> 2,,,,441,,441,,,1,1 >>>>>>> >>>>>>> Profiles: >>>>>>> Ops,Gateway,Domain,Subscriber,Profile Id,Rate Id1,00-H1,Rate >>>>>>> Id2,H1-H2,Rate Id3,H2-H3,Rate Id4,H3-24 >>>>>>> 2,,,,441,441,8,441,19,441,24,441,0 >>>>>>> >>>>>>> Rates: >>>>>>> Ops,Gateway,Domain,Subscriber,Rate Id,Destination,Price,App,Connect >>>>>>> 2,,,,441,131,2100,audio,0 >>>>>>> >>>>>>> U did not have a rate id and no App. Without App field for me it >>>>>>> does >>>>>>> not calculate either. >>>>>>> >>>>>>> DanB >>>>>>> >>>>>>>> >>>>>>>> But, the command ShowPrice returns: >>>>>>>> ShowPrice From=sip:101@sd-7501.dedibox.fr To=0033677832974 >>>>>>>> Duration=59 >>>>>>>> 0.0000 >>>>>>>> >>>>>>>> >>>>>>>> I have a field TrafficRate in the database but doesn't appear on >>>>>>>> the >>>>>>>> CDRTool interface >>>>>>>> I put random values inside >>>>>>>> >>>>>>>> >>>>>>>> Dan-Cristian Bogos a écrit : >>>>>>>>> My version is: >>>>>>>>> CDRTool version 5.1.3 >>>>>>>>> >>>>>>>>> In my rates table I have one more field more which is >>>>>>>>> TrafficRate. I >>>>>>>>> see that this one is missing on your side. >>>>>>>>> Can u also create a default rate table without filling any of the >>>>>>>>> gateway, subscriber, domain in profiles? This one should match all >>>>>>>>> the >>>>>>>>> destinations. Maybe would be useful till u make it work to have >>>>>>>>> only >>>>>>>>> the test row and remove anything else to take it gradually. >>>>>>>>> Also, try querying for price with 00 added in the front of the >>>>>>>>> destination since this is the default prefix set in CDRTool for >>>>>>>>> international destinations (without being necessary to be added in >>>>>>>>> rate or destination table). >>>>>>>>> >>>>>>>>> DanB >>>>>>>>> >>>>>>>>> On 8/9/07, Marc LEURENT <lftsy(a)free.fr> wrote: >>>>>>>>>> I have added: >>>>>>>>>> >>>>>>>>>> Destination: >>>>>>>>>> Ops,Gateway,Domain,Subscriber,Destination Id,Description >>>>>>>>>> 2,,sd-7501.dedibox.fr,,31800,Ma Dst TEST >>>>>>>>>> 2,,,33677832974,131,TEST INTERNATIONAL >>>>>>>>>> >>>>>>>>>> Profiles: >>>>>>>>>> Ops,Gateway,Domain,Subscriber,Profile Id,Rate Id1,00-H1,Rate >>>>>>>>>> Id2,H1-H2,Rate Id3,H2-H3,Rate Id4,H3-24 >>>>>>>>>> 2,,sd-7501.dedibox.fr,101(a)sd-7501.dedibox.fr,441,441,8,441,19,441,24,441,0 >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> 2,,sd-7501.dedibox.fr,106(a)sd-7501.dedibox.fr,441,441,8,441,19,441,24,441,0 >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> 2,,,33677832974,441,441,22,441,23,441,24,441,0 >>>>>>>>>> >>>>>>>>>> Rates: >>>>>>>>>> Ops,Gateway,Domain,Subscriber,Rate >>>>>>>>>> Id,Destination,Price,App,Connect >>>>>>>>>> 2,,sd-7501.dedibox.fr,,441,31800,2100,audio,4647 >>>>>>>>>> 2,,sd-7501.dedibox.fr,101(a)sd-7501.dedibox.fr,441,31800,200,audio,2364 >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> 2,,sd-7501.dedibox.fr,102(a)sd-7501.dedibox.fr,441,31800,200,audio,245 >>>>>>>>>> >>>>>>>>>> 2,,,33677832974,441,131,2100,,2426 >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> ShowPrice From=sip:101@sd-7501.dedibox.fr To=33677832974 >>>>>>>>>> Duration=59 >>>>>>>>>> 0.0000 >>>>>>>>>> >>>>>>>>>> But it is still not working... >>>>>>>>>> Could you spent some time sending me a part of yours... >>>>>>>>>> And are you using CDRTool 5.2.1?? >>>>>>>>>> >>>>>>>>>> Thanks >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> Dan-Cristian Bogos a écrit : >>>>>>>>>>> Hi Marc, >>>>>>>>>>> >>>>>>>>>>> can u do me a favor and try adding in an international >>>>>>>>>>> destination in >>>>>>>>>>> your pricelist and try placing a call to it. See if you get >>>>>>>>>>> billing >>>>>>>>>>> calculation for it. I suspect that your billing doesn't match >>>>>>>>>>> the >>>>>>>>>>> destination due to prefixes which are normally stripped. Not >>>>>>>>>>> quite a >>>>>>>>>>> CDRTool internals expert but my installation simply works for >>>>>>>>>>> international destinations. >>>>>>>>>>> >>>>>>>>>>> DanB >>>>>>>>>>> >>>>>>>>>>> On 8/9/07, Marc LEURENT <lftsy(a)free.fr> wrote: >>>>>>>>>>>> Hello Dan-Cristian, >>>>>>>>>>>> >>>>>>>>>>>> I just want a simple thing, that when I'm connecting to the >>>>>>>>>>>> rating engine using telnet: telnet 127.0.0.1 9024 >>>>>>>>>>>> When I enter such a command, that the price should be != 0: >>>>>>>>>>>> >>>>>>>>>>>> ShowPrice From=sip:101@sd-7501.dedibox.fr >>>>>>>>>>>> To=sip:106@sd-7501.dedibox.fr Duration=59 >>>>>>>>>>>> 0.0000 >>>>>>>>>>>> >>>>>>>>>>>> Help would be very grateful... >>>>>>>>>>>> Thanks... >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> The rating engine is requesting the good radaact200708 table (I >>>>>>>>>>>> have change my globac.inc conf file >>>>>>>>>>>> "ser_radius"=>array( >>>>>>>>>>>> "name" => "OpenSER", >>>>>>>>>>>> "class" => "CDRS_ser_radius", >>>>>>>>>>>> "table" => >>>>>>>>>>>> "radacct".date("Ym"), >>>>>>>>>>>> ...... >>>>>>>>>>>> >>>>>>>>>>>> ) >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> Here is a view of the rating tables... And I don't understand >>>>>>>>>>>> where I'm wrong.. >>>>>>>>>>>> >>>>>>>>>>>> Destinations: >>>>>>>>>>>> Ops,Gateway,Domain,Subscriber,Destination Id,Description >>>>>>>>>>>> 2,,sd-7501.dedibox.fr,,31800,Ma Dst TEST >>>>>>>>>>>> >>>>>>>>>>>> Customers: >>>>>>>>>>>> Ops,Gateway,Domain,Subscriber,WeekDay,Fallback,WeekEnd,Fallback,Timezone,Incr,Minim >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> 2,,sd-7501.dedibox.fr,101(a)sd-7501.dedibox.fr,441,,442,,Europe/Paris,20,1 >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> 2,,sd-7501.dedibox.fr,106(a)sd-7501.dedibox.fr,441,,442,,Europe/Paris,34,1 >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> Profiles: >>>>>>>>>>>> Ops,Gateway,Domain,Subscriber,Profile Id,Rate Id1,00-H1,Rate >>>>>>>>>>>> Id2,H1-H2,Rate Id3,H2-H3,Rate Id4,H3-24 >>>>>>>>>>>> 2,,sd-7501.dedibox.fr,,441,442,8,441,19,441,24,441,0 >>>>>>>>>>>> 2,,sd-7501.dedibox.fr,,442,442,24,442,0,442,0,442,0 >>>>>>>>>>>> >>>>>>>>>>>> Rates: >>>>>>>>>>>> Ops,Gateway,Domain,Subscriber,Rate >>>>>>>>>>>> Id,Destination,Price,App,Connect >>>>>>>>>>>> 2,,,,441,31800,2100,audio,0 >>>>>>>>>>>> 2,,,,442,31800,200,audio,0 >>>>>>>>>>>> >>>>>>>>>>>> Prepaid: >>>>>>>>>>>> Ops,Account,Balance,Last change,Lock,Start time,Last >>>>>>>>>>>> price,Max,Destination >>>>>>>>>>>> 2,101(a)sd-7501.dedibox.fr,10030.0000,2007-08-08 >>>>>>>>>>>> 17:57:00,,0000-00-00 00:00:00,0.0000,0, >>>>>>>>>>>> 2,106(a)sd-7501.dedibox.fr,20010.0000,2007-08-08 >>>>>>>>>>>> 17:38:09,,0000-00-00 00:00:00,0.0000,0, >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> ShowPrepaidAccounts Filter=sd-7501.dedibox.fr >>>>>>>>>>>> Account=101(a)sd-7501.dedibox.fr Balance=10030.0000 >>>>>>>>>>>> Account=106(a)sd-7501.dedibox.fr Balance=20010.0000 >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> And here is my radacct200708 table: >>>>>>>>>>>> RadAcctId AcctSessionId AcctUniqueId UserName >>>>>>>>>>>> Realm NASIPAddress NASPortId NASPortType >>>>>>>>>>>> AcctStartTime AcctStopTime AcctSessionTime >>>>>>>>>>>> AcctAuthentic ConnectInfo_start ConnectInfo_stop >>>>>>>>>>>> AcctInputOctets AcctOutputOctets CalledStationId >>>>>>>>>>>> CallingStationId AcctTerminateCause >>>>>>>>>>>> ServiceType FramedProtocol FramedIPAddress AcctStartDelay >>>>>>>>>>>> AcctStopDelay SipMethod SipResponseCode SipToTag >>>>>>>>>>>> SipFromTag >>>>>>>>>>>> SipTranslatedRequestURI SipUserAgents >>>>>>>>>>>> SipApplicationType SipCodecs SipRPID >>>>>>>>>>>> SipRPIDHeader SourceIP SourcePort CanonicalURI DelayTime >>>>>>>>>>>> Timestamp DestinationId Rate Price Normalized >>>>>>>>>>>> BillingId MediaInfo RTPStatistics FromHeader >>>>>>>>>>>> UserAgent Contact >>>>>>>>>>>> >>>>>>>>>>>> "2";"vjyxdmwkbbjajnc(a)192.168.95.50";"336224af44cd989a";"sip:106@sd-7501.dedibox.fr";"sd-7501.dedibox.fr";"88.191.45.91";"5060";;"2007-08-09 >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> 10:55:58";"2007-08-09 >>>>>>>>>>>> 10:56:03";"5";NULL;NULL;;"0";"0";;;"200";"Sip-Session";;;"0";"0";"INVITE";"200";"c0a80101-20e94fd";"nxitc";;;;;;;;;"101(a)sd-7501.dedibox.fr";;"0";;;NULL;"0";;NULL;;;; >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> And my syslog: >>>>>>>>>>>> >>>>>>>>>>>> Aug 9 11:00:03 sd-7501 CDRTool[17377]: Normalize datasource >>>>>>>>>>>> ser_radius, database DB_radius, table radacct200708 >>>>>>>>>>>> Aug 9 11:00:03 sd-7501 CDRTool[17377]: Lock >>>>>>>>>>>> ser_radius:radacct200708 >>>>>>>>>>>> Aug 9 11:00:03 sd-7501 CDRTool[17097]: ShowPrice Id=2 >>>>>>>>>>>> Timestamp=1186649758 Duration=5 inputTraffic=0 outputTraffic=0 >>>>>>>>>>>> From=sip:106@sd-7501.dedibox.fr >>>>>>>>>>>> To=101(a)sd-7501.dedibox.fr Gateway= Application=audio >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>> _______________________________________________ >>>>>> Users mailing list >>>>>> Users(a)openser.org >>>>>> http://openser.org/cgi-bin/mailman/listinfo/users >>> > _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

5 6

[Serusers] sip express router for windows
by Youngjin Park 15 Aug '07

15 Aug '07

Is there a sip express router for windows? If not, do you have any plan in near future? Thanks. Young

2 1

[Serusers] record_route() & record_route_preset() di
by Kostas Marneris 15 Aug '07

15 Aug '07

Hello, I have a question regarding record_route() & record_route_preset() functions. I faced up a problem with some SIP clients with inbound calls where I believe that it's related with these functions. 1/ When I use the "record_route()" function in a single server SER setup I have the following "Record-Route:" header in INVITE mesg from SER to called SIP UA: Record-Route: <sip:1.2.3.4;ftag=AAA-19972;lr=on> ... 2/ When I use the "record_route_preset()" function in a double server SER setup (2 SER servers behind a Cisco CSM Load balancer), example : record_route_preset("5.6.7.8"); I have the following "Record-Route:" header in INVITE mesg (SER -> called SIP UA): Record-Route: <sip:1234567890@5.6.7.8;ftag=XPH-5307;lr=on> ... In first case the SIP UA accepts the inbound call and everything is Ok. In second case the same SIP UA doesn't replied with '100 Trying' mesg back to SER. In the second case, SER also adds the called number "1234567890" in the Record-Route header. Why there is this difference and if that can cause problems (incompatibility) to some SIP clients ? thanks for any hint, Kostas

2 1

[Serusers] ser asterisk pstn gateway
by ronald＠stanford.edu 15 Aug '07

15 Aug '07

hello all, I am stuck and need your help for +xlite +ser +asterisk +pstn calls. I dial 914082743304 and get "caller Not Found". I am able to authenticate and complete sip -> sip calls, but cannot complete PSTN calls I did monitor asterisk and no reroute was offered. please help! permissions.allow: ^sip:91[0-9]{10}@bigu\.edu$ ^sip:9011[0-9]*@bigu\.edu$ ^sip:9[0-9]{10}@bigu\.edu$ ngrep -vvvv proto udp port 5060 produces the following response: 171.66.176.123:47729 -> 192.168.11.243:5060 INVITE sip:914082743304@bigu.edu SIP/2.0..Via: SIP/2.0/UDP 192.168.1.100:47729;br anch=z9hG4bK-d87543-4f5eb56fc40f1f43-1--d87543-;rport..Max-Forwards: 70..Contact: <si p:cell@192.168.1.100:47729>..To: "914082743304"<sip:914082743304@bigu.edu>..From: "SER EXpress"<sip:cell@bigu.edu>;tag=b9283f27..Call-ID: Y2JkY2ZmMDQyMjQ3YTljYTg5 NTdlOGFiNGUyZWVlY2Q...CSeq: 1 INVITE..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER , NOTIFY, MESSAGE, SUBSCRIBE, INFO..Content-Type: application/sdp..User-Agent: eyeBea m release 1011s stamp 41121..Content-Length: 421....v=0..o=- 1 2 IN IP4 192.168.1.100 ..s=CounterPath eyeBeam 1.5..c=IN IP4 192.168.1.100..t=0 0..m=audio 32942 RTP/AVP 107 100 106 6 0 105 8 18 3 5 101..a=fmtp:18 annexb=yes..a=fmtp:101 0-15..a=rtpmap:107 BV 32/16000..a=rtpmap:100 SPEEX/16000..a=rtpmap:106 SPEEX-FEC/16000..a=rtpmap:105 SPEEX- FEC/8000..a=rtpmap:18 G729/8000..a=rtpmap:101 telephone-event/8000..a=sendrecv..a=x-r tp-session-id:AEB31A44A04C4B32BD3D72349E4FBB80.. # U 192.168.11.243:5060 -> 171.66.176.123:47729 SIP/2.0 404 caller Not Found..Via: SIP/2.0/UDP 192.168.1.100:47729;branch=z9hG4bK-d875 43-4f5eb56fc40f1f43-1--d87543-;rport=47729;received=171.66.176.123..To: "914082743304 "<sip:914082743304@bigu.edu>;tag=21f9bfca23409407a263e4a7e40abe9d.f2a8..From: "SE R EXpress"<sip:cell@bigu.edu>;tag=b9283f27..Call-ID: Y2JkY2ZmMDQyMjQ3YTljYTg5NTdl OGFiNGUyZWVlY2Q...CSeq: 1 INVITE..Server: Sip EXpress router (0.9.4 (i386/linux))..Co ntent-Length: 0..Warning: 392 192.168.11.243:5060 "Noisy feedback tells: pid=15307 re q_src_ip=171.66.176.123 req_src_port=47729 in_uri=sip:914082743304@bigu.edu out_u ri=sip:914082743304@bigu.edu via_cnt==1".... debug=3 fork=yes log_stderror=no listen=1xx.x4.11.xx3 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 alias=sip2.bigu.edu 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/domain.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/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") #added these timers for voicemail forwarding modparam("tm", "fr_inv_timer", 25) modparam("tm", "noisy_ctimer", 1) 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("1xx.x4.11.xx3:5060;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 ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); break; }; if (method=="INVITE") { if (!allow_trusted()) { if (!proxy_authorize("sip2.bigu.edu","subscriber")) { proxy_challenge("sip2.bigu.edu","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; if (client_nat_test("3")||search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { sl_send_reply("403", "Forbidden"); }; break; }; if (method=="ACK") { route(1); break; } else if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (!is_uri_host_local()) { route(4); 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("sip2.bigu.edu","subscriber")) { www_challenge("sip2.bigu.edu","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (!allow_trusted()) { if (!proxy_authorize("sip2.bigu.edu","subscriber")) { proxy_challenge("sip2.bigu.edu","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; if (uri=~"^sip:1[0-9]{10}@bigu.edu") { strip(1); }; lookup("aliases"); if (!is_uri_host_local()) { route(4); route(1); break; }; if (uri=~"^sip:011[0-9]*@bigu.edu") { # International PSTN route(4); route(5); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@bigu.edu") { # Domestic PSTN route(4); route(5); break; }; sl_send_reply("404", "caller Not Found"); break; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { if (!isflagset(8)) { setflag(8); use_media_proxy(); }; }; } route[5] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("1xx.x4.8.xx8"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); 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(); }; }

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[OpenSER-Users] openser 1.2.2 scheduled for Thursday, August 16, 2007
by Daniel-Constantin Mierla 14 Aug '07

14 Aug '07

Hello, we plan a minor release of 1.2.2 for Thursday, August 16, 2007. It was discussed on the last irc meeting. If you know a critical bug that should stop this release, please let us know. The ones registered on the tracker will be reviewed and considered for final decision of 1.2.2, don't send them again. Cheers, Daniel

2 1

[OpenSER-Users] Saving public IP in CDR
by Andreas Sikkema 14 Aug '07

14 Aug '07

Hi, I'm trying to do add accounting to our OpenSER server. Getting accounting to work is easy but I'm having trouble saving client IP addresses with every CDR. Most of our SIP UAs are behind some form of NAT and to "prove" that calls have originated at the client I want to save the public IP address and as extra service also the private IP address of the UA. Saving the private IP address is easy, $Ri contains this value. I'm having trouble finding out what to do to save the public IP address. I thought that since I'm using fix_contact() I could save the contact header body, but $ct still contained the private IP address. Is this even possible? -- Andreas Sikkema

2 2

[OpenSER-Users] New H350 Module
by Christian Schlatter 14 Aug '07

14 Aug '07

Hello everybody! A new openser module named H350 is available in the openser devel repository. http://www.openser.org/docs/modules/devel/h350.html This module allows a script writer to implement H.350 based SIP digest authentication, call forwarding, SIP URI alias to AOR rewriting, and service level parsing. H.350 is an ITU-T Recommendation that standardizes how real-time communications (RTC) account data is stored in an LDAP directory. In particular, H.350.4 defines LDAP object classes for storing SIP account data. H.350 and H.350.4 are also described in IETF RFC 3944. The H350 module uses the LDAP module API to query H.350 directory servers, and it exports script functions to parse H.350 LDAP attributes into openser AVPs. This allows to integrate openser proxies with an enterprise/provider's existing identity management infrastructure. The module documentation should provide all information needed to use this module with your openser installation. Feel free to ask questions regarding this module on this list. /Christian - University of North Carolina

1 0

[OpenSER-Users] Click 2 Call / click To Call applications
by liran tal 14 Aug '07

14 Aug '07

Hey everyone, Following a very interesting but old and un-answered post here: http://www.voipuser.org/forum_topic_6616.html about the integration of Click-To-Call web applications with OpenSER I wanted to give it a try myself through the FIFO interface. Before jumping into scripts and code I was wondering if someone can make it clear how to use those fifo commands that "initiate" a call between 2 parties, from that post, the user uses: /* construct FIFO command */ $fifo_cmd=":t_uac:".$myfilename."\n". "REFER\n".$caller."\n". "p-version: ".$signature."\n". "Contact: ".$web_contact."\n". "Referred-By: ".$web_contact."\n". "Refer-To: ".$callee."\n". "\n". /* EoHeader */ ".\n\n"; /* EoFifoRequest */ Is there documentation available on the various fifo commands like this that I can use? Can I simply echo the fifo command string to /tmp/openser_fifo? Thanks in advance, Regards, Liran.

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[OpenSER-Users] Logging failed registration attempt
by Edoardo Serra 14 Aug '07

14 Aug '07

Hi all, I'd like to log failed SIP REGISTER attempt either with xlog or with sip_trace() but I cannot understand where to put related code to catch the authentication error Here is the part of my opensr.cfg dedicated to REGISTER handling if (method=="REGISTER") { if (!proxy_authorize("exorsa", "openser_view")) { proxy_challenge("exorsa", "0"); exit; } if (!check_to()) { sl_send_reply("403", "Digest username and URI username do NOT match! Stay away!"); exit; } save("location"); exit; }; Tnx in advance Regards Edoardo

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[OpenSER-Users] OpenSER coredumps on acc.so module
by liran tal 14 Aug '07

14 Aug '07

Hey everyone, I've enabled RADIUS accounting in my config file and setup radiusclient.confand the servers file with the appropriate data though OpenSER coredumps for some reason, this is the backtrace from gdb: #0 0xb7afc168 in test_config (rh=0x824d450, filename=0x824b240 "/var/csis/cfg/radiusclient.conf") at config.c:389 389 config.c: No such file or directory. in config.c (gdb) backtrace #0 0xb7afc168 in test_config (rh=0x824d450, filename=0x824b240 "/var/csis/cfg/radiusclient.conf") at config.c:389 #1 0xb7afbf49 in rc_read_config (filename=0x824b240 "/var/csis/cfg/radiusclient.conf") at config.c:312 #2 0xb7b1440b in extra2strar () from /usr/local/lib/openser/modules/acc.so #3 0x0808dab1 in init_child () #4 0x0806d15b in main () servers file: 192.168.0.1 test123 the servers file was before with a key of some non-standard chars like "test123!!5%^#" so I changed it to "test123" thinking that this might have caused it but this is not the reason. Thanks in advnace, Lir.

2 2

[OpenSER-Users] Werid records in 'presentity'table when PUBLISHing
by KevinKinnan 14 Aug '07

14 Aug '07

If the Route Header(sip:[ip add]:[port]) or Outbound Proxy(sip:[ip add]:[port]) in the PUBLISH request has no Lr parameter, the username field in 'presentity' table is null, and the domain field is also wrong. Without Lr parameter(no loose routing), the request received by Presence server is that placing the remote target URI into the Request URI(rfc 3561), and adding a Route Header with the value of the remote target URI. It seems the presence module just takes the Request-URI(which actually is the address of presence server) in the received request and save username and domain into the database. Thus, the username is null and the domain is the address of presence server. Is that right?? Why doesn't this happen in 1.2.0 version?THANK YOURegards,Kevin _________________________________________________________________ 通过 Windows Live Spaces 与朋友轻松共享您的生活。 http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx…

2 1

[OpenSER-Users] a=nortpproxy:yes in SDP and fix_nated_*
by liran tal 14 Aug '07

14 Aug '07

Hey everyone, When I'm doing sequential forking I'm seeing that for each gateway that doesn't answer and I move forward in the list of available gateways to forward the INVITE to, openser adds an "a=nortpproxy:yes" into the SDP so what happens is that the final gateway which is alive receives the sip header with multiple lines of "a=nortpproxy:yes" repeating themselves in the SDP and throws away the packet. >From my browsing in nathelper's code I see the only place that writes a "a=nortpproxy:yes" string to the SDP is the fix_nated_sdp function, and I'm actually not using it in my cfg but rather using fix_nated_register and fix_nated_contact which from what the README says are not handling the SDP part. And so I'm asking, where could it be coming from? And also, it would be easier if I can insert a piece of code that can do some regular expression miracles with the SIP header and remove any duplicate lines of the "nortpproxy" string. I'm running on openser 1.1.1 Thanks, Lir.

2 2

[Serusers] Out of Office
by chayn＠imhocorp.com 14 Aug '07

14 Aug '07

Hello, I am out of the office and will be unable to respond to messages until Thursday, August 16. Claude

1 0

[OpenSER-Users] [avp_db_query] How to analize resulting AVP's row by row?
by Iñaki Baz Castillo 14 Aug '07

14 Aug '07

Hi, in order to implement a ACL system for incoming calls I've created a table "acl" with those columns: - priority - from_username - from_domain - to_username - group_from - group_to - type | action In that tables I store rules, and my idea is to work as a firewall: the first match is ejecuted (action=ALLOW or action=REJECT) and exit. But I can't do it since I don't know how to use all the AVP's JUST for the first row, then for the second: avp_db_query("SELECT priority,from_username,from_domain,to_username,group_from,group_to,type,action FROM acl", "$avp(s:priority);$avp(s:from_username);$avp(s:from_domain);$avp(s:to_username);$avp(s:group_from);$avp(s:group_to);$avp(s:type);$avp(s:action)"); This query results in many rows, but I can't analize each row, I just can compare each AVP with a value, but for that AVP they are possible all the values of any row. Because this I'm thinking in using the "exec" module to call a system script. Is not possible to access to tjhe query result by rows? Thanks for any help. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

2 4

[OpenSER-Users] Failure Route - 408 Timeout & Asterisk
by Kelvin Williams 13 Aug '07

13 Aug '07

I would like to advise my users as to the reason the phone stopped ringing (no, I'm not going to let it ring for 20 minutes as many of them wish that I would). My grand design is to send the call to Asterisk which plays a recording to the effect of, "the party you have dialed does not answer, please try your call later" after the 2 minute timeout. Upon 408 the call is sent to Asterisk, however there is no audio. The call stays alive until Asterisk hits the Hangup() in the dialplan. I've tested by adding extra audio (to make sure the call does stay up) and watching on the Asterisk console. In other situations, audio passes fine between the Asterisk and the endpoint (ie. Calling voicemail). Any advice would be greatly appreciated. Some relevant portions of the config: loadmodule "tm.so" modparam("tm", "fr_timer", 3) modparam("tm", "fr_inv_timer", 120) modparam("tm", "wt_timer", 10) modparam("tm", "fr_inv_timer_avp", "$avp(s:timeout)") route[ROUTE_TO_PSTN] { .SNIP. (RURI fixes, etc.) rewritehostport("PSTN_IP:PSTN_PORT"); if (!is_method("BYE")) { t_on_reply("ROUTE_TO_PSTN_ONREPLY"); t_on_failure("ROUTE_TO_PSTN_FAILURE"); if (isflagset(FLAG_DEBUG)) xlog("L_INFO", "Relaying to PSTN - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci - On Reply, On Failure Set\n"); }; if (!t_relay()) { if (isflagset(FLAG_DEBUG)) xlog("L_INFO", "Relaying to PSTN - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci - Sending Reply Error\n"); sl_reply_error(); }; exit; } failure_route[ROUTE_TO_PSTN_FAILURE] { .SNIP. (SQL, etc for updating the CDR) if (t_check_status("408")) { rewriteuri("sip:ANC-408@ANS_IP:ANS_PORT"); append_branch(); t_relay(); exit; }; exit; }

1 0

[OpenSER-Users] remote IP address
by Victor Gamov 13 Aug '07

13 Aug '07

Hi All! Is it possible to know IP-address to which request will be forwarded ? I see that t_relay() actually know it but how get these knowledge in script? any pseudo-variable available for it? -- CU, Victor Gamov

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[OpenSER-Users] Dispatcher round robin....
by Francisco Ivandic 13 Aug '07

13 Aug '07

Hi All, I'm using dispatcher succesfully, with round robin alg (option 4), and also use ds_mark_dst and ds_next_dst to select other proxy in case of failure. It balances great when all servers are active, but when there are inactive servers, round robin alg selects the next in the round, even if it's marked as inactive. This results in going through ds_next_dst, and overloads the first active server in the dispatcher file until completing the round. Then, other active servers are called, and when passing to a group of inactive servers, starts going through the first active (as much times as inactive servers there are), and again completing the round. Does anyone know a way to implement a round robin aware of inactive servers? Thanks in advance, Francisco Ivandic

1 0

[OpenSER-Users] subs_status is still active after unsubscription, is it right?
by KevinKinnan 13 Aug '07

13 Aug '07

Thank your Anca for subscribe tip!Notify request with terminated subscription status is got when unsubscribing. But the subs_status in watchers table is still active, is that right? Regards,Kevin _________________________________________________________________ 通过 Windows Live Spaces 与朋友轻松共享您的生活。 http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx…

2 1

[OpenSER-Users] trunk version 2592: NOTIFY problem
by KevinKinnan 13 Aug '07

13 Aug '07

SUBSCRIBE request can not save in watchers table properly. Both p_user and w_user are subscriber's name!? In this case, they are both 'alice'. The SUBSCRIBE request is shown as follows:----------------------------------------SUBSCRIBE sip:alice@148.88.232.101 SIP/2.0Call-ID: c14d16bf8f758655b09588b764c3f0c9(a)148.88.232.101CSeq: 20 SUBSCRIBEFrom: "alice" <sip:alice@148.88.232.101:5061>;tag=z9hG4bK41052461To: "bob" <sip:bob@148.88.232.101:5062>Via: SIP/2.0/TCP 148.88.232.101:5061;branch=z9hG4bK57563Max-Forwards: 70Expires: 360000Event: presenceRoute: <sip:148.88.233.209:5065;lr>Contact: "alice" <sip:alice@148.88.232.101:5061>Content-Length: 0----------------------------------------HELP!Regards,Kevin _________________________________________________________________ 通过 Live.com 查看资讯、娱乐信息和您关心的其他信息！ http://www.live.com/getstarted.aspx

2 1

[Serusers] preload_udomain: bad callid
by Steffen Waldmann 13 Aug '07

13 Aug '07

hi there, when i restart my ser i get the following error: 0(0) preload_udomain: ERROR: bad callid record in table location 0(0) preload_udomain: ERROR: for username 1000, contact sip:1000@192.168.2.117:30316;rinstance=80910b76433a8887 0(0) preload_udomain: ERROR: skipping... 0(0) preload_udomain: ERROR: bad callid record in table location 0(0) preload_udomain: ERROR: for username 1001, contact sip:1001@192.168.2.220:5060;transport=udp 0(0) preload_udomain: ERROR: skipping... ser doesnt start up again... what could be the problem?

1 0

[OpenSER-Users] About openser+asterisk voicemail
by zze-WU Yin RD-MAPS-LAN 13 Aug '07

13 Aug '07

Hello everybody, I have a question aboute the configuration of openser. I want to add asterisk voicemail, but I know that asterisk use the database 'openser' in the mysql. But my system use a openser+freeradius+mysql and I don't use a database 'openser' but 'radius'. So I want to know if it is possible to do a voice mail in my system? Any advice is welcom. Thank you very much!!!

2 1

[OpenSER-Users] trunk version 2592: NOTIFY problem
by KevinKinnan 13 Aug '07

13 Aug '07

BTW, My sip client is programmed by Jain-NIST 1.2 _________________________________________________________________ 探索 Windows Vista 的世界 http://search.live.com/results.aspx?q=windows+vista&mkt=zh-cn&FORM=LIVSOP

1 0

[OpenSER-Users] trunk version 2592: NOTIFY problem
by KevinKinnan 13 Aug '07

13 Aug '07

Hi,I've upgraded to the latest trunk version, and still got this failure. Finally, I sort it out. The 1.2.0 version allows the Route Header without Lr parameter, but the latest version does not. If there(in the latest trunk) is no Lr parameter in the Route Header, the username column in 'presentity' table will be null, while the PUBLISH message is not different from the one in 1.2.0 version.Regards,Kevin _________________________________________________________________ 中国十大热门旅游景点 http://search.msn.com/results.aspx?q=%E4%B8%AD%E5%9B%BD%E5%8D%81%E5%A4%A7%E…

1 0

[OpenSER-Users] FreeBSD Bug - mi_xmlrpc
by Jorge Guntanis 13 Aug '07

13 Aug '07

OS: FreeBSD openser 6.2-RELEASE FreeBSD 6.2-RELEASE #0: Fri Jan 12 10:40:27 UTC 2007 root@dessler.cse.buffalo.edu:/usr/obj/usr/src/sys/GENERIC i386 OpenSer Version root@openser# openser -V version: openser 1.3.0-dev10-notls (i386/freebsd) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, 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 poll method support: poll, select, kqueue. svnrevision: 2:2588M @(#) $Id: main.c 2578 2007-08-07 15:17:07Z bogdan_iancu $ main.c compiled on 01:33:27 Aug 10 2007 with gcc 3.4.6 also tried with release 2327 having the exact same result. Dependencies Version: libxml2-2.6.29 xmlrpc-c-0.9.10_2 libwww-5.4.0_4 Config File: # Uncomment this if you want to use SQL database loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "mi_xmlrpc.so" loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" Result: root@openser# openser ERROR: load_module: could not open module </usr/local/lib/openser/modules/mi_xmlrpc.so>: /usr/local/lib/libxmlrpc_client.so.7: Undefined symbol "HTThroughLine" parse error (53,13-14): failed to load module ERROR: bad config file (1 errors) Has anyone seen this before, any idea on how to fix this? Any help is appreciated. Thanks Jorge

3 2

Re: [OpenSER-Users] Problems with fifo
by Javier Allende Astigarraga 13 Aug '07

13 Aug '07

this is my openser.cfg # # $Id: openser.cfg,v 1.6.2.1 2006/07/17 15:51:03 klaus_darilion Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes # Set to no to enter debugging mode log_stderror=no # (cmd line: -E) Set to yes to enter debugging mode check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" # # 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 = "/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/openser/modules/mysql.so" loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/usrloc.so" loadmodule "/usr/lib/openser/modules/registrar.so" loadmodule "/usr/lib/openser/modules/textops.so" #loadmodule "mi_fifo.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/openser/modules/auth.so" loadmodule "/usr/lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("mi_fifo","fifo_name","/tmp/openser_fifo") #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"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # 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 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("localhost", "subscriber")) { www_challenge("localhost", "0"); exit; }; save("location"); 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 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; } -- ===================================================== Legolas_Bilbao[ID2006][GKR] Dios creo un equipo Perfecto a los demas los lleno de extranjeros http://www.forosindicedonkey.com http://usuarios.lycos.es/ligaforo/ =====================================================

2 1

[OpenSER-Users] Problems with fifo
by Javier Allende Astigarraga 13 Aug '07

13 Aug '07

I have installed Openser,but if i exec Openserctrl moni appears this message Error : error opening Openser's fifo /tmp/openser_fifo make sure you have line 'fifo=.....' in your config if i do touch /tmp/openserfifo the last message disappears and appears the following one -e [cycle #:1; if constant make sure server lives] my openser.cfg file i want to use mysql commented modparam("usrloc","db_mode",0) uncommented - loadmodule "/usr/lib/openser/modules/mysql.so" - loadmodule "/usr/lib/openser/modules/auth.so" - loadmodule "/usr/lib/openser/modules/auth_db.so" - modparam("usrloc", "db_mode", 2) - modparam("auth", "calculate_ha1", yes) - modparam("auth_db", "password_column", "password") - if (!www_authorize("localhost", "subscriber")) { - www_challenge("locahost", "0"); - break; - }; what i'm doing wrong?? -- ===================================================== Legolas_Bilbao[ID2006][GKR] Dios creo un equipo Perfecto a los demas los lleno de extranjeros http://www.forosindicedonkey.com http://usuarios.lycos.es/ligaforo/ =====================================================

2 5

[OpenSER-Users] trunk version 2592: NOTIFY problem
by KevinKinnan 13 Aug '07

13 Aug '07

Everything is ok when using openser 1.2.0, such as SUBSCRIBE, NOTIFY(except unsubscribe, terminated expected) and etc. Updating to 2592 trunk version, got NOTIFY error.Case 1:UE connects to presence module directly, PUBLISH data saves in DB, but no username is shown in the column. After publishing two users (both null shown in 'username' column), subscrib each other. Sometimes it got NOTIFY request from presence module, sometimes not, it is not stable. And even it receives NOTIFY message, the xml data is from subscriberself(maybe because of null username, and both domains are the same IP addr). It is so weird!Case 2:UE->IMS Open Core->Openser Presence, version 1.2.0 works well, but not the latest 2592 trunk version. This time the username is shown in the column when publishing, I cant figure out the reason? However, still no xml data in NOTIFY message when subscribing. Another weird thing is, the subscriber username column and watchee uername column are both shown as subscriber's name! Any thoughts plz?Thanks,Kevin _________________________________________________________________ 中国十大热门旅游景点 http://search.msn.com/results.aspx?q=%E4%B8%AD%E5%9B%BD%E5%8D%81%E5%A4%A7%E…

2 1

[OpenSER-Users] xmpp/jabber IM
by TCB 11 Aug '07

11 Aug '07

I'm trying to setup openser with instant messaging , is this possible with radius as authentication back end? Any sample configs ? -- TC

2 1

[OpenSER-Users] Regarding SNMP and Statistics in SER
by Parminder.Nayyar＠relianceada.com 11 Aug '07

11 Aug '07

Hi friends I have been on the look out for someting related to performance counters or statistics in SER. If i browse through the code, I find, that there is #ifdef STAT macro defined, which is used if STATs are enabled while compilation. Now, when i try to do that, it looks for a SNMP module related files... which it doesnt find... On searching on internet i find a page saying SNMP is dead... Can any one help me in finding out how to enable statistics for SER? It is extremely urgent for me... please help .. :(( Thanks Parminder Nayyar The information contained in this e-mail message is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you should return it to the sender immediately. Please note that while we scan all e-mails for viruses we cannot guarantee that any e-mail is virus-free and accept no liability for any damage caused by any virus transmitted by this email.

2 1

[OpenSER-Users] Issue on MediaProxy Module
by Sergio Gutierrez 10 Aug '07

10 Aug '07

Hello all. Currently, we have deployed OpenSER using mediaProxy for natted users, and it has happened to us the following issue, which we consider would be worth to be had present: When MediaProxy server is not able to answer because of any problem (Overloading, Temporary unavailableness, etc), Openser gets frozen and it is not able to answer any request. In our setup, Openser is running on Sun Fire T2000, and MediaProxy was running on Sun Fire V240 (Dual Processor), both running Solaris 10. It seems that mediaproxy does not scale well on SMP hardware (As it was discussed in list these days), and it overload, but we confimed the issue related to MediaProxy timeout, and we saw that anyway, OpenSER got frozen. Thanks for your attention. Kind Regards. Sergio Gutiérrez.

1 0

[OpenSER-Users] Mysql Error
by Gerson Matiolli 10 Aug '07

10 Aug '07

Somebody knows what it is this message of error? Aug 10 15:55:43 localhost /usr/local/sbin/openser[16746]: submit_query: Unknown column 'src_user' in 'field list' Aug 10 15:55:43 localhost /usr/local/sbin/openser[16746]: db_insert: Error while submitting query Aug 10 15:55:43 localhost /usr/local/sbin/openser[16746]: ERROR:acc:acc_db_request: Error while inserting to database

2 2

[Serusers] Voicemail common access.
by dciepiel＠orange.pl 10 Aug '07

10 Aug '07

Hi, If someone is logged in external network, (examlpe Asterisk) how to dial into his voicemail box. The Rquest_uri will not be recognized by voicemail. How to make voicemail prompt for user name and password? cheers, Daniel

2 1

[OpenSER-Users] Saving public IP in CDR
by Adrian Georgescu 10 Aug '07

10 Aug '07

Yes, add $si to the list of extra Radius attributes logged in accounting, for example: modparam("acc", "radius_extra", "Sip-RPid=$avp(s:rpid); \ Source-IP=$si; \ Source-Port=$sp; \ Canonical-URI=$avp(can_uri); \ Billing-Party=$avp (billing_party); \ Divert-Reason=$avp (s:divert_reason); \ X-RTP-Stat=$avp (s:rtp_statistics); \ From-Header=$hdr(from); \ User-Agent=$hdr(user-agent); \ Contact=$hdr(contact); \ Event=$hdr(event); \ SIP-Proxy-IP=$avp (s:sip_proxy_ip)") Adrian >>> Andreas Sikkema andreas.sikkema at bbeyond.nl Thu Aug 9 17:39:12 CEST 2007 Previous message: [OpenSER-Users] [Openser Users]Differences between SIP Server andPBX Next message: [OpenSER-Users] Saving public IP in CDR Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] Hi, I'm trying to do add accounting to our OpenSER server. Getting accounting to work is easy but I'm having trouble saving client IP addresses with every CDR. Most of our SIP UAs are behind some form of NAT and to "prove" that calls have originated at the client I want to save the public IP address and as extra service also the private IP address of the UA. Saving the private IP address is easy, $Ri contains this value. I'm having trouble finding out what to do to save the public IP address. I thought that since I'm using fix_contact() I could save the contact header body, but $ct still contained the private IP address. Is this even possible? -- Andreas Sikkema

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Re: [Serusers] SER query...
by Piyush.Bansal＠relianceada.com 10 Aug '07

10 Aug '07

Hi, I have the ethereal traces for the load run on SER. The 180 Ringing received from remote end has the second via field present in it. Even then i'm getting the same error. One more thing I've observed in the code that the value of 2nd via is being checked at two places. One in receive.c and another is forward.c. The check in receive.c is passed but the same check got failed at one stage later (forward.c). regds, ----- Forwarded by Parminder Nayyar/RCOM/RelianceADA on 08/07/2007 03:16 PM ----- "Weiter Leiter" <bp4mls(a)googlemail.com> Sent by: serusers-bounces(a)lists.iptel.org 08/07/2007 12:05 AM To "Piyush.Bansal(a)relianceada.com" <Piyush.Bansal(a)relianceada.com> cc serusers(a)lists.iptel.org Subject Re: [Serusers] SER query... In case you're using SER as a proxy, when it receives a request from upstream the message must contain already a Via; as it forwards the request downstream, it adds another Via/Via body, pointing to itself. When the reply comes from downstream, it must, thus, contain at least 2 Vias. Otherwise, the error is printed (and could indicate a broken/misconfigured downstream element). WL. On 8/4/07, Piyush.Bansal(a)relianceada.com < Piyush.Bansal(a)relianceada.com> wrote: hi all, Currently i was trying to test SER under load conditions. At the load of just 10 simultaneous calls, the following error occured. "ERROR: forward_reply: no 2nd via found in reply" Note: Sometimes the error occured after 100-110 call and sometime it goes upto 1000 calls.. Can anybody help me in this issue.... regds, Piyush Bansal _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers -- "C is a language that combines all the elegance and power of assembly language with all the readability and maintainability of assembly language." _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

2 1

[Serusers] Regarding Performance Counters
by Parminder.Nayyar＠relianceada.com 10 Aug '07

10 Aug '07

Hello friends I am looking for a way out to record as to how many INVITE messages have been processed, how many calls got in voice, how many remained in just signalling... for estimating/analysing the performance counters or traffic on the SER. I had posted this question earlier also, but dint get nething However, I just came thru the module AVP in SER, in which it says, we can store the attribute value pair.. adn can define our own attributes too.. Which gives me an idea to create a variable say.. invite_count, and everytime INVITE request is processed thru SER, I can increment it. However, A major doubt i have is that.. if i can perform mathematical operations on the AVP??? Can anyone help in this?? and Am i going right?? or something better exists?? Please help... Thanks! Parminder Nayyar

2 1

[Serusers] ODP:Re: Why max_forwards=16?
by dciepiel＠orange.pl 10 Aug '07

10 Aug '07

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 >

2 1

[Serusers] Need Help!!!
by Kapil Goel 10 Aug '07

10 Aug '07

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

2 1

[Serusers] ODP:Re: Missing ACK and BYE in SIP communication.
by dciepiel＠orange.pl 10 Aug '07

10 Aug '07

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 >

1 0

[Serusers] Missing ACK and BYE in SIP communication.
by dciepiel＠orange.pl 10 Aug '07

10 Aug '07

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

2 1

[Serusers] Why max_forwards=16?
by dciepiel＠orange.pl 10 Aug '07

10 Aug '07

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

2 1

[Serusers] "Noisy feedbeck tells" in SIP message Trying.
by dciepiel＠orange.pl 10 Aug '07

10 Aug '07

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

2 1

[OpenSER-Users] Why if (uri==myself) is always true? (about openser call number pstn)
by zze-WU Yin RD-MAPS-LAN 10 Aug '07

10 Aug '07

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); }

3 2

[OpenSER-Users] A function to know if a user is registered?
by Iñaki Baz Castillo 10 Aug '07

10 Aug '07

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

2 3

[Serusers] RPM spec file for ser_ctl
by Bruce Bauman 10 Aug '07

10 Aug '07

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

2 1

[Serusers] SER+PATTON1+PATTON2+PATTON3
by pokerino76 10 Aug '07

10 Aug '07

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.

2 1

[OpenSER-Users] [Openser Users]Differences between SIP Server and PBX
by Javier Allende Astigarraga 09 Aug '07

09 Aug '07

hi guys what are the differences between SIP servers and PBX?? thanks for all -- ===================================================== Legolas_Bilbao[ID2006][GKR] Dios creo un equipo Perfecto a los demas los lleno de extranjeros http://www.forosindicedonkey.com http://usuarios.lycos.es/ligaforo/ =====================================================

3 2

[Serusers] WG: SER+ PATTON1 + PATTON2
by pokerino76 09 Aug '07

09 Aug '07

thanks for your help, all work correctly.

1 0

Re: [OpenSER-Users] Re: How can it be simpler and working? CDRTool
by Marc LEURENT 09 Aug '07

09 Aug '07

YEAHHHHHH! My problem was with the destination database. I didn't understand what was the destination ID, I thought it was just a random one! I was probably too tired... Hopefully, tomorrow; it's summer holidays I owe you a beer!! Are you going to Rome next September for VoIP conference? ShowPrice From=sip:101@sd-7501.dedibox.fr To=sip:001305465768 Duration=599876 199.9587 App: audio Destination: 1305 (TEST INTERNATIONAL) Customer: default Connect fee: 0.0000 -- Span: 1 Duration: 11931 s StartTime: 2007-08-09 15:41:09 (Europe/Paris) ProfileId: 441 for weekday RateId: 441 for 8-19h Rate: 0.0200 / 60 s Price: 3.9770 -- Span: 2 Duration: 29931 s StartTime: 2007-08-09 15:41:09 (Europe/Paris) ProfileId: 441 for weekday RateId: 442 for 19-24h Rate: 0.0200 / 60 s Price: 9.9770 -- Span: 3 Duration: 58731 s StartTime: 2007-08-09 15:41:09 (Europe/Paris) ProfileId: 441 for weekday RateId: 442 for 0-8h Rate: 0.0200 / 60 s Price: 19.5770 Yes: "reportMissingRates"=> 1 flavio a écrit : > Marc, > > Is the missing rates parameter of $CDRTool['rating'] equal to 1? > > rates.csv > Ops,Gateway,Domain,Subscriber,Rate Id,Destination,Price,App,Connect > 2,,192.168.1.160,,442,1305,200,audio, > 2,,192.168.1.160,,441,1305,200,audio, Ops,Gateway,Domain,Subscriber,Rate Id,Destination,Price,App,Connect 2,,sd-7501.dedibox.fr,,442,1305,200,audio, 2,,sd-7501.dedibox.fr,,441,1305,200,audio, > > > profiles.csv > Ops,Gateway,Domain,Subscriber,Profile Id,Rate Id1,00-H1,Rate > Id2,H1-H2,Rate Id3,H2-H3,Rate Id4,H3-24 > 2,,192.168.1.160,,441,442,8,441,19,442,24,,0 > 2,,192.168.1.160,,442,442,24,442,19,441,24,,0 Ops,Gateway,Domain,Subscriber,Profile Id,Rate Id1,00-H1,Rate Id2,H1-H2,Rate Id3,H2-H3,Rate Id4,H3-24 2,,sd-7501.dedibox.fr,,442,442,24,442,19,442,24,,0 2,,sd-7501.dedibox.fr,,441,442,8,441,19,442,24,,0 > > > customer.csv > > Ops,Gateway,Domain,Subscriber,WeekDay,Fallback,WeekEnd,Fallback,Timezone,Incr,Minim > > 2,,,,441,,442,,Europe/Amsterdam,0,0 Ops,Gateway,Domain,Subscriber,WeekDay,Fallback,WeekEnd,Fallback,Timezone,Incr,Minim 2,,,,441,,442,,Europe/Paris,0,0 > > > I have made all the tests dialing 001305...... (Miami). (I have used > 192.168.1.160 as the domain, change to yourdomain). It normalizes to > 1305 striping two zeros as specified on cdrlib.phtml. Check if it is > striping the digits to match the desired destination. > > class CDRS { > > var $CDR_class = 'CDR'; > var $intAccessCode = '00'; > var $natAccessCode = '0'; > class CDRS { var $CDR_class = 'CDR'; var $intAccessCode = '00'; var $natAccessCode = '0'; var $RatingEngineSocket = false; var $maxrowsperpage = 15; var $status = array(); var $normalizedField = 'Normalized'; var $DestinationIdField = 'DestinationId'; var $BillingIdField = 'UserName'; var $sipTraceDataSource = 'sip_trace'; var $defaults = array(); var $whereUnnormalized = ''; ...................... ...................... > Cheers, > > Flavio > ----- Original Message ----- From: "Marc LEURENT" <lftsy(a)free.fr> > To: "flavio" <flavio(a)asteriskguide.com> > Sent: Thursday, August 09, 2007 9:41 AM > Subject: Re: [OpenSER-Users] Re: How can it be simpler and working? CDRTool > > >> Hello flavio, >> I would be very grateful if you can send me just the .csv of your >> tables about your internal calls, the simpliest one for you that is >> working... >> Best Regards! >> >> Marc, desperate >> >> Marc LEURENT a écrit : >>> I've got a running tail -f /var/lig/messages and I haven't seen a >>> single error... >>> It's just that it calculates 0.0000 for the price >>> >>> I have already added the $CDRTool['rating'] parameter.. >>> >>> Id don't have the same message: >>> Aug 9 14:00:02 sd-7501 CDRTool[18946]: ShowPrice Id=3 >>> Timestamp=1186660750 Duration=11 inputTraffic=0 outputTraffic=0 >>> From=sip:106@sd-7501.dedibox.fr >>> To=6713(a)sd-7501.dedibox.fr Gateway= Application=audio >>> Aug 9 14:00:02 sd-7501 CDRTool[19627]: Normalization done in 0 s, >>> memory usage: 9.90 MB >>> Aug 9 14:00:02 sd-7501 CDRTool[19627]: 1 CDRs, 1 normalized in 0 s @ >>> 0 cps >>> >>> But there is no "ConnectFee=0.0000 Span=1 Duration=57 De >>> stId=1305 domain=192.168.1.160 Profile=441 Period=weekday Rate=441 >>> Interval=8 >>> -19 Cost=0.0200/10 Price=0.1140 " after >>> >>> Do I have to activate something else?? >>> >>> >>> flavio a écrit : >>>> Marc, >>>> >>>> I'm using 5.1.5 and it is working. Check the /var/log/syslog for errors >>>> in the normalization process >>>> >>>> Something like: >>>> >>>> Aug 6 07:18:55 debian CDRTool[3973]: Error: Cannot find rates for >>>> callid=, d >>>> omain=192.168.1.160, destination=1305, profile=441, app=audio >>>> >>>> The correct in the tests I have made is: >>>> >>>> Aug 6 07:47:44 debian CDRTool[3973]: ShowPrice Id=88 >>>> Timestamp=1186400781 Du >>>> ration=57 inputTraffic=0 outputTraffic=0 From=1000(a)192.168.1.160 >>>> To=001305345 >>>> 6789(a)192.168.1.160 Gateway=192.168.1.118 Application=audio >>>> Aug 6 07:47:44 debian CDRTool[3973]: ConnectFee=0.0000 Span=1 >>>> Duration=57 De >>>> stId=1305 domain=192.168.1.160 Profile=441 Period=weekday Rate=441 >>>> Interval=8 >>>> -19 Cost=0.0200/10 Price=0.1140 >>>> >>>> please activate the following parameter in the global.inc file >>>> (reportMissingRates). >>>> >>>> $CDRTool['rating']=array("priceDenominator" => >>>> 10000,"priceDecimalDigits" => 4,"minimumDurationCharged"=> >>>> 0,"durationPeriodRated"=> 60,"trafficSizeRated"=> >>>> 1024,"reportMissingRates"=> 1,"minimumDuration"=> 0); >>>> >>>> I hope it helps, it took me a long time before to start working. >>>> >>>> Cheers >>>> >>>> Flavio >>>> >>>> >>>> ----- Original Message ----- From: "Marc LEURENT" <lftsy(a)free.fr> >>>> To: "flavio" <flavio(a)asteriskguide.com> >>>> Cc: <users(a)openser.org> >>>> Sent: Thursday, August 09, 2007 9:01 AM >>>> Subject: Re: [OpenSER-Users] Re: How can it be simpler and working? >>>> CDRTool >>>> >>>> >>>>> Yes, I've added all rights to my account... >>>>> and price is NULL in the database until a Normalize is done! All >>>>> prices are set to 0.0000 after... >>>>> >>>>> And after a >>>>> Normalize datasource ser_radius, database DB_radius, table >>>>> radacct200708 >>>>> Aug 9 14:00:02 sd-7501 CDRTool[18946]: ShowPrice Id=3 >>>>> Timestamp=1186660750 Duration=11 inputTraffic=0 outputTraffic=0 >>>>> From=sip:106@sd-7501.dedibox.fr >>>>> To=6713(a)sd-7501.dedibox.fr Gateway= Application=audio >>>>> >>>>> >>>>> >>>>> >>>>> flavio a écrit : >>>>>> Marc, >>>>>> >>>>>> Have you checked your user rights in the CDRTool (is show price item >>>>>> highlighted?). My CDRtool shown x.xxx in the price and that's was >>>>>> the >>>>>> cause. >>>>>> >>>>>> Flavio >>>>>> >>>>>> ----- Original Message ----- From: "Marc LEURENT" <lftsy(a)free.fr> >>>>>> To: "Dan-Cristian Bogos" <dan.bogos(a)gmail.com> >>>>>> Cc: <users(a)openser.org> >>>>>> Sent: Thursday, August 09, 2007 8:01 AM >>>>>> Subject: [OpenSER-Users] Re: How can it be simpler and working? >>>>>> CDRTool >>>>>> >>>>>> >>>>>> I'm starting to fell sorry to bother you like that... >>>>>> I have changed the database like you said, but it is still not >>>>>> working... >>>>>> >>>>>> ShowPrice From=sip:101@sd-7501.dedibox.fr To=0033677832974 >>>>>> Duration=5923 >>>>>> 0.0000 >>>>>> >>>>>> Should I try to downgrade to version 5.1.3 like yours? >>>>>> >>>>>> ShowMemory >>>>>> 1 destinations >>>>>> 1 customers >>>>>> 1 profiles >>>>>> 1 rates >>>>>> 0 holidays >>>>>> 2 prepaid accounts >>>>>> 1 sessions >>>>>> 1 sessions last minute >>>>>> 1 sessions last hour >>>>>> 1 sessions last day >>>>>> >>>>>> >>>>>> but in the database, the prepaid account is credited by 10.00 >>>>>> ShowPrepaidAccounts Filter=101(a)sd-7501.dedibox.fr >>>>>> Account=101(a)sd-7501.dedibox.fr Balance=10040.0000 >>>>>> >>>>>> ShowCustomers >>>>>> default >>>>>> >>>>>> >>>>>> ShowAccounts >>>>>> Error: Invalid request >>>>>> >>>>>> MaxSessionTime From=sip:101@sd-7501.dedibox.fr >>>>>> To=sip:106@sd-7501.dedibox.fr Duration=7200 Lock=1 >>>>>> none >>>>>> >>>>>> DebitBalance From=sip:101@sd-7501.dedibox.fr >>>>>> To=sip:106@sd-7501.dedibox.fr Duration=59 >>>>>> Ok >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Dan-Cristian Bogos a écrit : >>>>>>> Ok, let's take it in your way. I would rather modify your data into: >>>>>>> >>>>>>> Destinations: >>>>>>> Ops,Gateway,Domain,Subscriber,Destination Id,Description >>>>>>> 2,,,,131,TEST INTERNATIONAL >>>>>>> >>>>>>> Customers: >>>>>>> Ops,Gateway,Domain,Subscriber,WeekDay,Fallback,WeekEnd,Fallback,Timezone,Incr,Minim >>>>>>> >>>>>>> >>>>>>> >>>>>>> 2,,,,441,,441,,,1,1 >>>>>>> >>>>>>> Profiles: >>>>>>> Ops,Gateway,Domain,Subscriber,Profile Id,Rate Id1,00-H1,Rate >>>>>>> Id2,H1-H2,Rate Id3,H2-H3,Rate Id4,H3-24 >>>>>>> 2,,,,441,441,8,441,19,441,24,441,0 >>>>>>> >>>>>>> Rates: >>>>>>> Ops,Gateway,Domain,Subscriber,Rate Id,Destination,Price,App,Connect >>>>>>> 2,,,,441,131,2100,audio,0 >>>>>>> >>>>>>> U did not have a rate id and no App. Without App field for me it >>>>>>> does >>>>>>> not calculate either. >>>>>>> >>>>>>> DanB >>>>>>> >>>>>>>> >>>>>>>> But, the command ShowPrice returns: >>>>>>>> ShowPrice From=sip:101@sd-7501.dedibox.fr To=0033677832974 >>>>>>>> Duration=59 >>>>>>>> 0.0000 >>>>>>>> >>>>>>>> >>>>>>>> I have a field TrafficRate in the database but doesn't appear on >>>>>>>> the >>>>>>>> CDRTool interface >>>>>>>> I put random values inside >>>>>>>> >>>>>>>> >>>>>>>> Dan-Cristian Bogos a écrit : >>>>>>>>> My version is: >>>>>>>>> CDRTool version 5.1.3 >>>>>>>>> >>>>>>>>> In my rates table I have one more field more which is >>>>>>>>> TrafficRate. I >>>>>>>>> see that this one is missing on your side. >>>>>>>>> Can u also create a default rate table without filling any of the >>>>>>>>> gateway, subscriber, domain in profiles? This one should match all >>>>>>>>> the >>>>>>>>> destinations. Maybe would be useful till u make it work to have >>>>>>>>> only >>>>>>>>> the test row and remove anything else to take it gradually. >>>>>>>>> Also, try querying for price with 00 added in the front of the >>>>>>>>> destination since this is the default prefix set in CDRTool for >>>>>>>>> international destinations (without being necessary to be added in >>>>>>>>> rate or destination table). >>>>>>>>> >>>>>>>>> DanB >>>>>>>>> >>>>>>>>> On 8/9/07, Marc LEURENT <lftsy(a)free.fr> wrote: >>>>>>>>>> I have added: >>>>>>>>>> >>>>>>>>>> Destination: >>>>>>>>>> Ops,Gateway,Domain,Subscriber,Destination Id,Description >>>>>>>>>> 2,,sd-7501.dedibox.fr,,31800,Ma Dst TEST >>>>>>>>>> 2,,,33677832974,131,TEST INTERNATIONAL >>>>>>>>>> >>>>>>>>>> Profiles: >>>>>>>>>> Ops,Gateway,Domain,Subscriber,Profile Id,Rate Id1,00-H1,Rate >>>>>>>>>> Id2,H1-H2,Rate Id3,H2-H3,Rate Id4,H3-24 >>>>>>>>>> 2,,sd-7501.dedibox.fr,101(a)sd-7501.dedibox.fr,441,441,8,441,19,441,24,441,0 >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> 2,,sd-7501.dedibox.fr,106(a)sd-7501.dedibox.fr,441,441,8,441,19,441,24,441,0 >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> 2,,,33677832974,441,441,22,441,23,441,24,441,0 >>>>>>>>>> >>>>>>>>>> Rates: >>>>>>>>>> Ops,Gateway,Domain,Subscriber,Rate >>>>>>>>>> Id,Destination,Price,App,Connect >>>>>>>>>> 2,,sd-7501.dedibox.fr,,441,31800,2100,audio,4647 >>>>>>>>>> 2,,sd-7501.dedibox.fr,101(a)sd-7501.dedibox.fr,441,31800,200,audio,2364 >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> 2,,sd-7501.dedibox.fr,102(a)sd-7501.dedibox.fr,441,31800,200,audio,245 >>>>>>>>>> >>>>>>>>>> 2,,,33677832974,441,131,2100,,2426 >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> ShowPrice From=sip:101@sd-7501.dedibox.fr To=33677832974 >>>>>>>>>> Duration=59 >>>>>>>>>> 0.0000 >>>>>>>>>> >>>>>>>>>> But it is still not working... >>>>>>>>>> Could you spent some time sending me a part of yours... >>>>>>>>>> And are you using CDRTool 5.2.1?? >>>>>>>>>> >>>>>>>>>> Thanks >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> Dan-Cristian Bogos a écrit : >>>>>>>>>>> Hi Marc, >>>>>>>>>>> >>>>>>>>>>> can u do me a favor and try adding in an international >>>>>>>>>>> destination in >>>>>>>>>>> your pricelist and try placing a call to it. See if you get >>>>>>>>>>> billing >>>>>>>>>>> calculation for it. I suspect that your billing doesn't match >>>>>>>>>>> the >>>>>>>>>>> destination due to prefixes which are normally stripped. Not >>>>>>>>>>> quite a >>>>>>>>>>> CDRTool internals expert but my installation simply works for >>>>>>>>>>> international destinations. >>>>>>>>>>> >>>>>>>>>>> DanB >>>>>>>>>>> >>>>>>>>>>> On 8/9/07, Marc LEURENT <lftsy(a)free.fr> wrote: >>>>>>>>>>>> Hello Dan-Cristian, >>>>>>>>>>>> >>>>>>>>>>>> I just want a simple thing, that when I'm connecting to the >>>>>>>>>>>> rating engine using telnet: telnet 127.0.0.1 9024 >>>>>>>>>>>> When I enter such a command, that the price should be != 0: >>>>>>>>>>>> >>>>>>>>>>>> ShowPrice From=sip:101@sd-7501.dedibox.fr >>>>>>>>>>>> To=sip:106@sd-7501.dedibox.fr Duration=59 >>>>>>>>>>>> 0.0000 >>>>>>>>>>>> >>>>>>>>>>>> Help would be very grateful... >>>>>>>>>>>> Thanks... >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> The rating engine is requesting the good radaact200708 table (I >>>>>>>>>>>> have change my globac.inc conf file >>>>>>>>>>>> "ser_radius"=>array( >>>>>>>>>>>> "name" => "OpenSER", >>>>>>>>>>>> "class" => "CDRS_ser_radius", >>>>>>>>>>>> "table" => >>>>>>>>>>>> "radacct".date("Ym"), >>>>>>>>>>>> ...... >>>>>>>>>>>> >>>>>>>>>>>> ) >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> Here is a view of the rating tables... And I don't understand >>>>>>>>>>>> where I'm wrong.. >>>>>>>>>>>> >>>>>>>>>>>> Destinations: >>>>>>>>>>>> Ops,Gateway,Domain,Subscriber,Destination Id,Description >>>>>>>>>>>> 2,,sd-7501.dedibox.fr,,31800,Ma Dst TEST >>>>>>>>>>>> >>>>>>>>>>>> Customers: >>>>>>>>>>>> Ops,Gateway,Domain,Subscriber,WeekDay,Fallback,WeekEnd,Fallback,Timezone,Incr,Minim >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> 2,,sd-7501.dedibox.fr,101(a)sd-7501.dedibox.fr,441,,442,,Europe/Paris,20,1 >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> 2,,sd-7501.dedibox.fr,106(a)sd-7501.dedibox.fr,441,,442,,Europe/Paris,34,1 >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> Profiles: >>>>>>>>>>>> Ops,Gateway,Domain,Subscriber,Profile Id,Rate Id1,00-H1,Rate >>>>>>>>>>>> Id2,H1-H2,Rate Id3,H2-H3,Rate Id4,H3-24 >>>>>>>>>>>> 2,,sd-7501.dedibox.fr,,441,442,8,441,19,441,24,441,0 >>>>>>>>>>>> 2,,sd-7501.dedibox.fr,,442,442,24,442,0,442,0,442,0 >>>>>>>>>>>> >>>>>>>>>>>> Rates: >>>>>>>>>>>> Ops,Gateway,Domain,Subscriber,Rate >>>>>>>>>>>> Id,Destination,Price,App,Connect >>>>>>>>>>>> 2,,,,441,31800,2100,audio,0 >>>>>>>>>>>> 2,,,,442,31800,200,audio,0 >>>>>>>>>>>> >>>>>>>>>>>> Prepaid: >>>>>>>>>>>> Ops,Account,Balance,Last change,Lock,Start time,Last >>>>>>>>>>>> price,Max,Destination >>>>>>>>>>>> 2,101(a)sd-7501.dedibox.fr,10030.0000,2007-08-08 >>>>>>>>>>>> 17:57:00,,0000-00-00 00:00:00,0.0000,0, >>>>>>>>>>>> 2,106(a)sd-7501.dedibox.fr,20010.0000,2007-08-08 >>>>>>>>>>>> 17:38:09,,0000-00-00 00:00:00,0.0000,0, >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> ShowPrepaidAccounts Filter=sd-7501.dedibox.fr >>>>>>>>>>>> Account=101(a)sd-7501.dedibox.fr Balance=10030.0000 >>>>>>>>>>>> Account=106(a)sd-7501.dedibox.fr Balance=20010.0000 >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> And here is my radacct200708 table: >>>>>>>>>>>> RadAcctId AcctSessionId AcctUniqueId UserName >>>>>>>>>>>> Realm NASIPAddress NASPortId NASPortType >>>>>>>>>>>> AcctStartTime AcctStopTime AcctSessionTime >>>>>>>>>>>> AcctAuthentic ConnectInfo_start ConnectInfo_stop >>>>>>>>>>>> AcctInputOctets AcctOutputOctets CalledStationId >>>>>>>>>>>> CallingStationId AcctTerminateCause >>>>>>>>>>>> ServiceType FramedProtocol FramedIPAddress AcctStartDelay >>>>>>>>>>>> AcctStopDelay SipMethod SipResponseCode SipToTag >>>>>>>>>>>> SipFromTag >>>>>>>>>>>> SipTranslatedRequestURI SipUserAgents >>>>>>>>>>>> SipApplicationType SipCodecs SipRPID >>>>>>>>>>>> SipRPIDHeader SourceIP SourcePort CanonicalURI DelayTime >>>>>>>>>>>> Timestamp DestinationId Rate Price Normalized >>>>>>>>>>>> BillingId MediaInfo RTPStatistics FromHeader >>>>>>>>>>>> UserAgent Contact >>>>>>>>>>>> >>>>>>>>>>>> "2";"vjyxdmwkbbjajnc(a)192.168.95.50";"336224af44cd989a";"sip:106@sd-7501.dedibox.fr";"sd-7501.dedibox.fr";"88.191.45.91";"5060";;"2007-08-09 >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> 10:55:58";"2007-08-09 >>>>>>>>>>>> 10:56:03";"5";NULL;NULL;;"0";"0";;;"200";"Sip-Session";;;"0";"0";"INVITE";"200";"c0a80101-20e94fd";"nxitc";;;;;;;;;"101(a)sd-7501.dedibox.fr";;"0";;;NULL;"0";;NULL;;;; >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> And my syslog: >>>>>>>>>>>> >>>>>>>>>>>> Aug 9 11:00:03 sd-7501 CDRTool[17377]: Normalize datasource >>>>>>>>>>>> ser_radius, database DB_radius, table radacct200708 >>>>>>>>>>>> Aug 9 11:00:03 sd-7501 CDRTool[17377]: Lock >>>>>>>>>>>> ser_radius:radacct200708 >>>>>>>>>>>> Aug 9 11:00:03 sd-7501 CDRTool[17097]: ShowPrice Id=2 >>>>>>>>>>>> Timestamp=1186649758 Duration=5 inputTraffic=0 outputTraffic=0 >>>>>>>>>>>> From=sip:106@sd-7501.dedibox.fr >>>>>>>>>>>> To=101(a)sd-7501.dedibox.fr Gateway= Application=audio >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>> _______________________________________________ >>>>>> Users mailing list >>>>>> Users(a)openser.org >>>>>> http://openser.org/cgi-bin/mailman/listinfo/users >>> >

1 0

[Serusers] Record-route
by dciepiel＠orange.pl 09 Aug '07

09 Aug '07

Hi, Does anybody know how to set up the record-route parameter to public adres in ser.cfg flie or else? Thanks for any help.\ cheers, Daniel

2 1

[OpenSER-Users] OpenSER + CDRTool
by Marc LEURENT 09 Aug '07

09 Aug '07

Hello Dan-Cristian, I just want a simple thing, that when I'm connecting to the rating engine using telnet: telnet 127.0.0.1 9024 When I enter such a command, that the price should be != 0: ShowPrice From=sip:101@sd-7501.dedibox.fr To=sip:106@sd-7501.dedibox.fr Duration=59 0.0000 Help would be very grateful... Thanks... The rating engine is requesting the good radaact200708 table (I have change my globac.inc conf file "ser_radius"=>array( "name" => "OpenSER", "class" => "CDRS_ser_radius", "table" => "radacct".date("Ym"), ...... ) Here is a view of the rating tables... And I don't understand where I'm wrong.. Destinations: Ops,Gateway,Domain,Subscriber,Destination Id,Description 2,,sd-7501.dedibox.fr,,31800,Ma Dst TEST Customers: Ops,Gateway,Domain,Subscriber,WeekDay,Fallback,WeekEnd,Fallback,Timezone,Incr,Minim 2,,sd-7501.dedibox.fr,101(a)sd-7501.dedibox.fr,441,,442,,Europe/Paris,20,1 2,,sd-7501.dedibox.fr,106(a)sd-7501.dedibox.fr,441,,442,,Europe/Paris,34,1 Profiles: Ops,Gateway,Domain,Subscriber,Profile Id,Rate Id1,00-H1,Rate Id2,H1-H2,Rate Id3,H2-H3,Rate Id4,H3-24 2,,sd-7501.dedibox.fr,,441,442,8,441,19,441,24,441,0 2,,sd-7501.dedibox.fr,,442,442,24,442,0,442,0,442,0 Rates: Ops,Gateway,Domain,Subscriber,Rate Id,Destination,Price,App,Connect 2,,,,441,31800,2100,audio,0 2,,,,442,31800,200,audio,0 Prepaid: Ops,Account,Balance,Last change,Lock,Start time,Last price,Max,Destination 2,101(a)sd-7501.dedibox.fr,10030.0000,2007-08-08 17:57:00,,0000-00-00 00:00:00,0.0000,0, 2,106(a)sd-7501.dedibox.fr,20010.0000,2007-08-08 17:38:09,,0000-00-00 00:00:00,0.0000,0, ShowPrepaidAccounts Filter=sd-7501.dedibox.fr Account=101(a)sd-7501.dedibox.fr Balance=10030.0000 Account=106(a)sd-7501.dedibox.fr Balance=20010.0000 And here is my radacct200708 table: RadAcctId AcctSessionId AcctUniqueId UserName Realm NASIPAddress NASPortId NASPortType AcctStartTime AcctStopTime AcctSessionTime AcctAuthentic ConnectInfo_start ConnectInfo_stop AcctInputOctets AcctOutputOctets CalledStationId CallingStationId AcctTerminateCause ServiceType FramedProtocol FramedIPAddress AcctStartDelay AcctStopDelay SipMethod SipResponseCode SipToTag SipFromTag SipTranslatedRequestURI SipUserAgents SipApplicationType SipCodecs SipRPID SipRPIDHeader SourceIP SourcePort CanonicalURI DelayTime Timestamp DestinationId Rate Price Normalized BillingId MediaInfo RTPStatistics FromHeader UserAgent Contact "2";"vjyxdmwkbbjajnc(a)192.168.95.50";"336224af44cd989a";"sip:106@sd-7501.dedibox.fr";"sd-7501.dedibox.fr";"88.191.45.91";"5060";;"2007-08-09 10:55:58";"2007-08-09 10:56:03";"5";NULL;NULL;;"0";"0";;;"200";"Sip-Session";;;"0";"0";"INVITE";"200";"c0a80101-20e94fd";"nxitc";;;;;;;;;"101(a)sd-7501.dedibox.fr";;"0";;;NULL;"0";;NULL;;;; And my syslog: Aug 9 11:00:03 sd-7501 CDRTool[17377]: Normalize datasource ser_radius, database DB_radius, table radacct200708 Aug 9 11:00:03 sd-7501 CDRTool[17377]: Lock ser_radius:radacct200708 Aug 9 11:00:03 sd-7501 CDRTool[17097]: ShowPrice Id=2 Timestamp=1186649758 Duration=5 inputTraffic=0 outputTraffic=0 From=sip:106@sd-7501.dedibox.fr To=101(a)sd-7501.dedibox.fr Gateway= Application=audio

3 12

[OpenSER-Users] Why if (uri==myself) is always true? (about openser call a number PSTN)
by zze-WU Yin RD-MAPS-LAN 09 Aug '07

09 Aug '07

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 # ------------------ 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); }; if (!method=="REGISTER") record_route(); if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); 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 (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] { 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] { 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); }

3 2

[OpenSER-Users] uri transformation errors in 1.2
by Andreas Granig 09 Aug '07

09 Aug '07

Hi, At the top of my config I've put the following snippet for some SEMS routings: if(uri =~ ";my_domain=.+") { $var(mydom) = $(ru{uri.param,my_domain}); xlog("L_INFO", "domain='$var(mydom)'\n"); $ru = "sip:" + $rU + "@" + $var(mydom); } It does nothing more than changing an r-uri "sip:user@domain;my_domain=otherdomain" to "sip:user@otherdomain". This works fine for some requests, but then it starts to complain: xl_get_ruri: ERROR while parsing the R-URI domain='' ERROR: parse_uri: bad host in uri (error at char ERROR: parse_sip_msg_uri: bad uri <sip:user@> So obviously the transformation $(ru{uri.param,my_domain}) doesn't return the parameter anymore, but it's present in the r-uri. After a restart it works again for some requests, then again stops working. Any hints? Andreas

3 8

[OpenSER-Users] Different Domains
by Danilo 09 Aug '07

09 Aug '07

Hi all, I am using Openser 1.2.1 with presence module. My question: Is possible to work with different domains in the presence module? For example: I have two domains: domain1 and domain2. Its "talk" among itselves. Did you understand? Thanks. Danilo

2 1

[OpenSER-Users] Conference
by amit 09 Aug '07

09 Aug '07

Hi All, How we can done conference in SIP ? We was already see in Tech-invite site but it not helpfully for us........ Please tell what we do for conference...... Thanks in advance, Amit Vijayvargiya

5 5

RE: [OpenSER-Users] Support for strict routing
by Vikas CG. 09 Aug '07

09 Aug '07

Thanks Dan for your reply, One more thing I want to know can we configure openser to be only complaint with RFC 2543 for testing purpose it should support RFC 3261. Thanks, Vikas C.G. -----Original Message----- From: Dan-Cristian Bogos [mailto:dan.bogos@gmail.com] Sent: Wednesday, August 08, 2007 6:02 PM To: Vikas CG. Cc: users(a)openser.org Subject: Re: [OpenSER-Users] Support for strict routing Hi there Vikas, OpenSER supports both types of routing, strict and loose-routing. It is just a matter of how you decide to handle your request inside routing scripts. In case of strict-routing, it is about keeping the RURI intact and let it flow out without calling loose-routing functions. Cheers, DanB On 8/8/07, Vikas CG. <vikas_cg(a)spanservices.com> wrote: > Hello everyone, > Can any body know whether openser supports for strict routing. > And how can we configure it. > Thanks, > Vikas C.G. > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

2 1

[Serusers] WG: SER+ PATTON1 + PATTON2
by Johannes Wagner-Meingassner 08 Aug '07

08 Aug '07

_______________________________ Hi! On the pattons you can create Error Messages when no circuit channel is available And on Ser you have to register a Failure Route to the route of the first patton and the failure route point to the second Patton. ----------------------------------------------------------- PATTON1: profile sip default map cause to-sip no-circuit-channel-available to 486 map cause to-sip resources-unavailable to 486 service hunt-group HUNT-BRI cyclic timeout 1 drop-cause normal-unspecified drop-cause no-circuit-channel-available drop-cause network-out-of-order drop-cause temporary-failure drop-cause switching-equipment-congestion drop-cause access-info-discarded drop-cause circuit-channel-not-available drop-cause resources-unavailable route call 1 dest-interface ISDN_01 route call 2 dest-interface ISDN_02 route call 3 dest-interface ISDN_03 --------------------------------------------------- SER: Ser.cfg: if ( uri=~".*sip:012345" ) # Patton 1 { t_on_failure("1"); t_relay_to_udp("11.11.11.10", "5060"); break; } --------------------------------------------------- failure_route[1] { if (method=="INVITE") { if (t_check_status ("600|486")) #### ErrorCode from Patton 1 { #####call from Patton 1 -- 486 (Busy) ret -- call switch to Patton 2 ###### if ( uri=~".*sip:012345" ) # Costumer xy { append_branch(); t_relay_to_udp("11.11.11.11", "5060"); ##Unterroute ist hier nicht notwendig, break; ##da sich die Verrechnung hier nicht aendert } } } --------------------------------------- I hope it help! Hannes _______________________________ Johannes Wagner-Meingassner -----Ursprüngliche Nachricht----- Von: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] Im Auftrag von pokerino76 Gesendet: Mittwoch, 8. August 2007 18:43 An: serusers Betreff: [Serusers] SER+ PATTON1 + PATTON2 Wichtigkeit: Hoch I have 2 patton from 8 channels everyone. I must make the sum of the channels 8+8=16. The patton they are connects to with the SER. How to make the switch of the channels when one of the two is saturate? _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers __________________________________________________________________________________ Dieses Mail wurde vom Infotech SecureMail Service ueberprueft und fuer sicher befunden. Fuer weitere Informationen zu Infotech SecureMail Service waehlen Sie bitte: www.infotech.at This email has been scanned by Infotech SecureMail Service and it has been classified as secure. For more information on Infotech SecureMail direct your web browser to: www.infotech.at

1 0

[OpenSER-Users] Issue calling use_media_proxy() at on_reply_route.
by Sergio Gutierrez 08 Aug '07

08 Aug '07

Hello all. We are facing some problems in our deployment of OpenSER, related to NAT Traversal solution using MediaProxy. In our scenario, using the the sample configuration NAT detection works right for REGISTER method, and for INVITE's sent from clients behind NAT, but, incoming calls originated from transparent clients, directed to clients behind NAT do not work, and do not pass through MediaProxy. Services based on re-INVITES do not work either. To solve that issue, we tried to intercept the 200 OK answer coming from NAT client, so that we could call use_media_proxy(), but, we get the error: "Empty Response from media_proxy()" Our configuration is more or less like this: # In the beginning of the main route if(client_nat_test("7")) { setflag(2); fix_contact(); }; ..... if(is_method("INVITE")) { if(isflagset(2)) { use_media_proxy(); }; t_on_reply("1"); } .... on_reply_route[1] { if (status =~ "(18[0-3])|(2[0-9][0-9])|(1[0-9][0-9])") { if (client_nat_test("3")) { fix_contact(); use_media_proxy(); } } } That is a very brief configuration fragment, but it is more or less the schema of configuration we are using. So far, our solution to be able to have the whole services working is marking all the calls so that they pass through the mediaproxy, but we are not sure if that is a good solution. Thanks in advance for every hint or suggestion. Kind regards *Sergio Armando Gutiérrez Betancur* *S**ubdirección **A**plicaciones **de **S**ervicios **C**onvergentes** UNE - EPM Telecomunicaciones * *Tel: 054 3802956* *Email: saguti(a)gmail.com*

3 4

[Serusers] SER+ PATTON1 + PATTON2
by pokerino76 08 Aug '07

08 Aug '07

I have 2 patton from 8 channels everyone. I must make the sum of the channels 8+8=16. The patton they are connects to with the SER. How to make the switch of the channels when one of the two is saturate?

1 0

[OpenSER-Users] RE: NAT question
by Papadopoulos Georgios 08 Aug '07

08 Aug '07

Hi Inaki, Why would you leave out the "4" flag test in nat_uac_test() for the replies? Of course if I leave out the "2" test then my example works (I have tested it) but I am afraid that this will break some other NATed senario. And if leave out the "2" test, should I also leave out the "16" test? Best regards George > ------------------------------ > > Message: 4 > Date: Wed, 8 Aug 2007 11:53:46 +0200 > From: I?aki Baz Castillo <ibc(a)in.ilimit.es> > Subject: Re: [OpenSER-Users] NAT question > To: users(a)openser.org > Message-ID: <200708081153.46621.ibc(a)in.ilimit.es> > Content-Type: text/plain; charset="iso-8859-1" > > El Wednesday 08 August 2007 10:30:54 Papadopoulos Georgios escribiσ: > > Hi Bogdan, > > > > I understand that 200 OK should not be fixed by proxyA. Maybe what I > > don't understand is what nat_uac_test("2") does and when to use it. > > > > Following is an example where > > clientA = demo1(a)altecnet.gr > > clientB = sip_test_1(a)i-call.gr > > proxyA = 213.5.43.4 > > proxyB = 213.5.43.134 > > > > When proxyA goes in the onreply_route for the 200 OK, > nat_uac_test("23") > > returns true (I guess because of the "2" flag) and proxyA fixes the > > contact again which is wrong. So, would it be safe to use > > nat_uac_test("23") for the requests and nat_uac_test("21") for the > > replies? I am afraid this would break the simple case with 2 NATed > > clients and only one proxy: > > clientA ------> proxyA ------> clientC > > > Good point. As I think, the correct NAT test for replies is: > nat_uac_test("1") > so we just check if <Contact:> header does contain an RFC1918 > IP address > (private IP address). > > In your case it will work since the <Contact> modified by > proxyB will contain > a non private IP so the NAT check (nat_uac_test("1")) in > proxyA wont detect > the repliy as behind NAT. > > Maybe others test could be done, but I think they should no > valid test that > just compare the received IP with the IP of the contact since > both could be > public IP but different, in which case shouldn't be > applied "fix_nat_contact". > > > Regards. > > > > -- > Iρaki Baz Castillo > ibc(a)in.ilimit.es > Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

2 1

Re: [Serusers] ODP:Re: ODP:Re: Instant messaging.
by tzieleniewski 08 Aug '07

08 Aug '07

to do this you need to substitute the value of the From Header with anonymous. try to use textops module cheers tomasz dciepiel(a)orange.pl pisze: > I mean, to hide the number the person who is calling, like in GSM. U > can hide the number and the person who answering your call doesn`t see > who is calling. See the attachment. > > cheers, > > Daniel > > > ---Oryginalna wiadomość--- > Nadawca:tzieleniewski <tzieleniewski(a)o2.pl> > Do:dciepiel@orange.pl > Otrzymana:2007-08-08 13:43 > Temat:Re: [Serusers] ODP:Re: Instant messaging. > do you mean call blocking at SIP Proxy?? > > dciepiel(a)orange.pl pisze: > >> Thanx Tomasz, It is very helpful. I have one more question. Do you >> know how to implement CLIR (Calling Line Identification Restriction) >> service? >> >> cheers, >> >> Daniel >> >> ---Oryginalna wiadomość--- >> Nadawca:tzieleniewski <tzieleniewski(a)o2.pl> >> Do:dciepiel@orange.pl >> Otrzymana:2007-08-08 11:18 >> Temat:Re: [Serusers] Instant messaging. >> use msilo module for this >> >> cheers tomasz >> >> dciepiel(a)orange.pl pisze: >> >> >>> Hi, >>> >>> I have a short question. Does SER support Instant messaging? If yes, I >>> will be gratefull for any quidelines how to set up it. >>> >>> >>> cheers, >>> >>> Daniel >>> >>> >>> > ------------------------------------------------------------------------ > >>> _______________________________________________ >>> Serusers mailing list >>> Serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >>> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >> > > > > ------------------------------------------------------------------------ >

1 0

[OpenSER-Users] Support for strict routing
by Vikas CG. 08 Aug '07

08 Aug '07

Hello everyone, Can any body know whether openser supports for strict routing. And how can we configure it. Thanks, Vikas C.G.

2 2

[Serusers] ODP:Re: Instant messaging.
by dciepiel＠orange.pl 08 Aug '07

08 Aug '07

Thanx Tomasz, It is very helpful. I have one more question. Do you know how to implement CLIR (Calling Line Identification Restriction) service? cheers, Daniel ---Oryginalna wiadomość--- Nadawca:tzieleniewski <tzieleniewski(a)o2.pl> Do:dciepiel@orange.pl Otrzymana:2007-08-08 11:18 Temat:Re: [Serusers] Instant messaging. use msilo module for this cheers tomasz dciepiel(a)orange.pl pisze: > Hi, > > I have a short question. Does SER support Instant messaging? If yes, I > will be gratefull for any quidelines how to set up it. > > > cheers, > > Daniel > > ------------------------------------------------------------------------ > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

2 1

RE: [OpenSER-Users] NAT question
by Papadopoulos Georgios 08 Aug '07

08 Aug '07

Hi Bogdan, I understand that 200 OK should not be fixed by proxyA. Maybe what I don't understand is what nat_uac_test("2") does and when to use it. Following is an example where clientA = demo1(a)altecnet.gr clientB = sip_test_1(a)i-call.gr proxyA = 213.5.43.4 proxyB = 213.5.43.134 When proxyA goes in the onreply_route for the 200 OK, nat_uac_test("23") returns true (I guess because of the "2" flag) and proxyA fixes the contact again which is wrong. So, would it be safe to use nat_uac_test("23") for the requests and nat_uac_test("21") for the replies? I am afraid this would break the simple case with 2 NATed clients and only one proxy: clientA ------> proxyA ------> clientC thank you for your help George # U 2007/08/08 10:11:34.233164 213.5.43.134:5060 -> 213.5.43.4:5060 SIP/2.0 200 Ok. Via: SIP/2.0/UDP 213.5.43.4;rport=5060;branch=z9hG4bK322.eea78413.0. Via: SIP/2.0/UDP 192.168.1.39;received=213.5.17.226;rport=5060;branch=z9hG4bKc0a801270000 002c46b96cbb000000a000000087. Record-Route: <sip:213.5.43.134;lr=on;ftag=109517115931019>. Record-Route: <sip:213.5.43.4;lr=on;ftag=109517115931019>. From: "222" <sip:demo1@altecnet.gr>;tag=109517115931019. To: <sip:sip_test_1@i-call.gr>;tag=0r8u9ziv3g. Call-ID: 000029DD-5407-0000-3452-00006A3B0000(a)192.168.1.39. CSeq: 2 INVITE. Contact: <sip:sip_test_1@213.5.17.236:5060;transport=udp;line=9yt55o5b>. User-Agent: snom190-3.56w. Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO. Allow-Events: talk, hold, refer. Supported: timer, 100rel, replaces. Content-Type: application/sdp. Content-Length: 208. . v=0. o=root 1592588598 1592588598 IN IP4 213.5.17.236. s=call. c=IN IP4 213.5.17.236. t=0 0. m=audio 10116 RTP/AVP 8 101. a=rtpmap:8 pcma/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. # U 2007/08/08 10:11:34.234013 213.5.43.4:5060 -> 213.5.17.226:5060 SIP/2.0 200 Ok. Via: SIP/2.0/UDP 192.168.1.39;received=213.5.17.226;rport=5060;branch=z9hG4bKc0a801270000 002c46b96cbb000000a000000087. Record-Route: <sip:213.5.43.134;lr=on;ftag=109517115931019>. Record-Route: <sip:213.5.43.4;lr=on;ftag=109517115931019>. From: "222" <sip:demo1@altecnet.gr>;tag=109517115931019. To: <sip:sip_test_1@i-call.gr>;tag=0r8u9ziv3g. Call-ID: 000029DD-5407-0000-3452-00006A3B0000(a)192.168.1.39. CSeq: 2 INVITE. Contact: <sip:sip_test_1@213.5.43.134:5060;transport=udp;line=9yt55o5b>. User-Agent: snom190-3.56w. Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO. Allow-Events: talk, hold, refer. Supported: timer, 100rel, replaces. Content-Type: application/sdp. Content-Length: 208. P-hint: NATed reply. . v=0. o=root 1592588598 1592588598 IN IP4 213.5.17.236. s=call. c=IN IP4 213.5.17.236. t=0 0. m=audio 10116 RTP/AVP 8 101. a=rtpmap:8 pcma/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Wednesday, August 08, 2007 9:50 AM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [OpenSER-Users] NAT question > > Hi George, > > each proxy should fix what it detects. ProxyA will take care > of the traffic related to A (since it detects A as NATed) and > ProxyB should take care of traffic for B (as it sees B as > nated). So, the 200 OK must be fixed only by proxyB. > > Regards, > bogdan > > Papadopoulos Georgios wrote: > > Hello all, > > > > I need some advice from all you NAT experts. Assume that we > have two > > OpenSER proxies serving two different domains and they both use > > nathelper. Also assume that their configuration is > identical and looks > > roughly like: > > > > route { > > ... > > record_route(); > > > > if (nat_uac_test("23")) { > > if (method!="REGISTER" && > > !is_present_hf("Record-Route")) { > > fix_nated_contact(); > > } > > } > > > > t_on_reply("1"); > > } > > > > onreply_route[1] { > > if (nat_uac_test("23")) { > > fix_nated_contact(); > > }; > > } > > clientA and clientB are both behind NAT. clientA@domainA > > <mailto:clientA@domainA> sends an INVITE to clientB@domainB > > <mailto:clientB@domainB> through proxyA: > > > > clientA@domainA <mailto:clientA@domainA> -----> ProxyA > -----> ProxyB > > ------> clientB@domainB <mailto:clientB@domainB> > > > > ProxyA detects that clientA is behind NAT and fixes the contact. > > ProxyB finds the Record-Route header, so it does not fix > the contact. > > My question is about how to handle the 200 OK. With this > configuration > > both proxies detect that clientB is behind NAT and the contact gets > > fixed twice. Is there a way to avoid this? > > > > thank you > > > > George > > > > > > > > > > > > > > Disclaimer > > > > The information in this e-mail and any attachments is > confidential. It > > is intended solely for the attention and use of the named > > addressee(s). If you are not the intended recipient, or person > > responsible for delivering this information to the intended > recipient, > > please notify the sender immediately. Unless you are the intended > > recipient or his/her representative you are not authorized to, and > > must not, read, copy, distribute, use or retain this message or any > > part of it. E-mail transmission cannot be guaranteed to be > secure or > > error-free as information could be intercepted, corrupted, lost, > > destroyed, arrive late or incomplete, or contain viruses. > > > > > ---------------------------------------------------------------------- > > -- > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > >

2 1

[Serusers] message counters
by Piyush.Bansal＠relianceada.com 08 Aug '07

08 Aug '07

Hi all, Is there any way in SER thru which i can maintain the message counters (Performance Counters), i.e. counts of each and every request and response...

1 0

[Serusers] Instant messaging.
by dciepiel＠orange.pl 08 Aug '07

08 Aug '07

Hi, I have a short question. Does SER support Instant messaging? If yes, I will be gratefull for any quidelines how to set up it. cheers, Daniel

3 2

[OpenSER-Users] OpenSER Oracle Connection
by Martin.Gruber1＠infineon.com 08 Aug '07

08 Aug '07

Hi @all, I want to run OpenSER with ORACLE Database. I use unixODBC and libsqora.so.10.1 to connect to the DB. But now I have problems with inserting Data in the DB. When OpenSER wants to add Data in the "location" table I always get an error because of the timeformat (expires, last_modified) I use SQL*Plus Client in Windows and when I write the Connection String from OpenSER usrloc here, everything is running well, Because I did alter session set nls_date_format = 'YYYY-MM-DD HH24:MM:SS'; Is it a Driver problem? Does anybody have a solution or experience with it? Thanks Martin

1 0

[OpenSER-Users] NAT question
by Papadopoulos Georgios 08 Aug '07

08 Aug '07

Hello all, I need some advice from all you NAT experts. Assume that we have two OpenSER proxies serving two different domains and they both use nathelper. Also assume that their configuration is identical and looks roughly like: route { ... record_route(); if (nat_uac_test("23")) { if (method!="REGISTER" && !is_present_hf("Record-Route")) { fix_nated_contact(); } } t_on_reply("1"); } onreply_route[1] { if (nat_uac_test("23")) { fix_nated_contact(); }; } clientA and clientB are both behind NAT. clientA@domainA sends an INVITE to clientB@domainB through proxyA: clientA@domainA -----> ProxyA -----> ProxyB ------> clientB@domainB ProxyA detects that clientA is behind NAT and fixes the contact. ProxyB finds the Record-Route header, so it does not fix the contact. My question is about how to handle the 200 OK. With this configuration both proxies detect that clientB is behind NAT and the contact gets fixed twice. Is there a way to avoid this? thank you George Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

2 1

[OpenSER-Users] SerWEB on OpenSER 1.2.x
by flavio 08 Aug '07

08 Aug '07

Hi, Has anyone got to install SerWEB with openser 1.2.x? I have tried php4, php5, xml-rpc, fifo. The browser freezes when logging in. If FIFO is not accessible it logs in, indicating a problem, maybe in the new mi_fifo module. Cheers, Flavio E. Goncalves

3 3

[OpenSER-Users] Openser Crashed
by M.Emran 08 Aug '07

08 Aug '07

i m using openser with postgresql.my openser crashed 5 times a day.anyonecan help me on this pls? -- Regards ---------- M Emran InSpiration Software Ltd.

2 1

[OpenSER-Users] radius+openser, OpenSer startup problem
by ram 08 Aug '07

08 Aug '07

Hi iam integrating with openser+radius iam able to test radius with authentication when iam running openser -v -f site.cfg ( config downloaded from http://www.openser.org/docs/openser-radius-1.0.x.html) iam getting the following error http://www.pastebin.ca/648858 any help ram

3 8

[Serusers] Blocking outgoing and incomming calls in SER.
by dciepiel＠orange.pl 07 Aug '07

07 Aug '07

Hi, Does anybody know how to set up an outgoing and incomming call blocking for specyfic subscriber? cheers, Daniel

1 0

[OpenSER-Users] SIP redirect
by Marc LEURENT 07 Aug '07

07 Aug '07

I just want to redirect all communication to phones 2xx to another SIP proxy through TCP. I have 2 problems, the first one is that the extensions don't exist in the location database, so I have added them manualy: openserctl ul add 299 sip:299@62.210.194.120 And when I use: if (method==INVITE && uri=~"sip:2[0-9][0-9]@*") { rewritehostport("62.210.194.120:5060"); t_relay("tcp:62.210.194.120:5060"); exit; }; I get: 0(17174) rewrite_uri: Rewriting Request-URI with 'sip:299@62.210.194.120' 0(17174) parse_headers: flags=ffffffffffffffff 0(17174) !!!! STARTING ROUTE 1: NORMAL FORWARD 0(17174) subst_run: running. r=1 0(17174) subst_str: no match 0(17174) !!!! STARTING ROUTE 3: SET/UNSET RTPPROXY 0(17174) DEBUG: t_newtran: T on entrance=0xffffffff 0(17174) parse_headers: flags=ffffffffffffffff 0(17174) parse_headers: flags=78 0(17174) t_lookup_request: start searching: hash=5349, isACK=1 Any idea?

1 1

Re: [OpenSER-Users] OpenSER + CDRTool
by Marc LEURENT 07 Aug '07

07 Aug '07

Yes I have followed the tutorial and after some changes due to upgrades, it's working, in the cdrtool web interface, I can see the cdrs... With the time of the calls, the canonical URI, the billing_party... But the price is not calculated... Is there any way to see if cdrtool is doing something? In my last email, i have posted the databases... Thanks Dan-Cristian Bogos a écrit : > Hello Marc, > > did u follow ./doc/INSTALL.txt? Can u access CDRTool via the web > interface? Make sure u can see the cdrs with the search tool. After > that it is a question of proper setting the pricelist (since it can be > pretty complex) and making sure u can connect to rating engine. > > Cheers, > DanB > > On 8/7/07, Marc LEURENT <lftsy(a)free.fr> wrote: >> Hello everybody, >> I was wondering what should I add in my cdrtool configuration to enable the communication price calculation >> CDRTool has evrything he needs from the freeradius.acct table, billing party and Canonical URI... >> >> >> Is it possible to monitor what CDRTool is doing??? >> Is there logs somewhere... >> Thanks >> >> >> >> I have also filled the billing_rates database: >> id gateway domain subscriber name destination durationRate trafficRate application connectCost >> "1";;"sd-7501.dedibox.fr";;;"103(a)sd-7501.dedibox.fr";"23";;"3";"2" >> >> >> destinations database: >> id gateway domain subscriber dest_id dest_name >> "1";;"sd-7501.dedibox.fr";"103(a)sd-7501.dedibox.fr";"1";"Domaine SIP locale" >> >> billing_customers: >> id gateway domain subscriber profile_name1 profile_name1_alt profile_name2 profile_name2_alt timezone increment min_duration country_code >> "1";;"sd-7501.dedibox.fr";"103(a)sd-7501.dedibox.fr";;;;;;"10";"10"; >> "2";;"sd-7501.dedibox.fr";"106(a)sd-7501.dedibox.fr";;;;;;"20";"35"; >> >> >> >> Here is what I see with CDRTool interface: >> >> 1N 2007-08-07 10:38:52 88.191.45.91 103(a)sd-7501.dedibox.fr 00:04 0.00 0.00 Ok (200) >> Signalling information >> Session: sjozozxmfficcar(a)192.168.95.47 >> Start time: 2007-08-07 10:38:52 Europe/Paris >> Stop time: 2007-08-07 10:38:56 Europe/Paris >> Method: INVITE from : >> From: >> Dialed URI: >> Canonical URI: 103(a)sd-7501.dedibox.fr >> Next hop URI: >> Billing Party: sip:106@sd-7501.dedibox.fr >> >> Rating information >> Free call >> >> >> >> Best Regards, >> >> Marc LEURENT >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >>

2 1

[OpenSER-Users] OpenSER + CDRTool
by Marc LEURENT 07 Aug '07

07 Aug '07

Hello everybody, I was wondering what should I add in my cdrtool configuration to enable the communication price calculation CDRTool has evrything he needs from the freeradius.acct table, billing party and Canonical URI... Is it possible to monitor what CDRTool is doing??? Is there logs somewhere... Thanks I have also filled the billing_rates database: id gateway domain subscriber name destination durationRate trafficRate application connectCost "1";;"sd-7501.dedibox.fr";;;"103(a)sd-7501.dedibox.fr";"23";;"3";"2" destinations database: id gateway domain subscriber dest_id dest_name "1";;"sd-7501.dedibox.fr";"103(a)sd-7501.dedibox.fr";"1";"Domaine SIP locale" billing_customers: id gateway domain subscriber profile_name1 profile_name1_alt profile_name2 profile_name2_alt timezone increment min_duration country_code "1";;"sd-7501.dedibox.fr";"103(a)sd-7501.dedibox.fr";;;;;;"10";"10"; "2";;"sd-7501.dedibox.fr";"106(a)sd-7501.dedibox.fr";;;;;;"20";"35"; Here is what I see with CDRTool interface: 1N 2007-08-07 10:38:52 88.191.45.91 103(a)sd-7501.dedibox.fr 00:04 0.00 0.00 Ok (200) Signalling information Session: sjozozxmfficcar(a)192.168.95.47 Start time: 2007-08-07 10:38:52 Europe/Paris Stop time: 2007-08-07 10:38:56 Europe/Paris Method: INVITE from : From: Dialed URI: Canonical URI: 103(a)sd-7501.dedibox.fr Next hop URI: Billing Party: sip:106@sd-7501.dedibox.fr Rating information Free call Best Regards, Marc LEURENT

1 0

[Serusers] drop and brake in ser.cfg
by tzieleniewski 07 Aug '07

07 Aug '07

Hi, Is there any difference in the brake and drop command behavior in ser.cfg?? Cheers Tomasz

4 5

[OpenSER-Users] Sending SIP from OpenSER to Asterisk boxes
by Arya 07 Aug '07

07 Aug '07

Hello Everyone Is it possible to have a OpenSER server in front of couple of Asterisk servers, than all SIP connects to OpenSER than OpenSER sends it to Asterisk boxes with load balancing? Thank You

2 1

[Serusers] problem with sercmd module
by dizel247＠aol.com 06 Aug '07

06 Aug '07

Hello, I have downloaded and intalled ser v2 from http://www.iptel.org/ser_2_0_0_release_candidate_1 Followd all the instructions. In Section Creating your first config. Run /usr/local/sbin/sercmd and type ctl.listen to see if SER is running and listening. Type help to see other commands available. I get an error. [root@localhost serweb-0.9.6]# /usr/local/sbin/ser -f /usr/local/etc/ser/ser.cfg Listening on ???????????? udp: 127.0.0.1 [127.0.0.1]:5060 ???????????? udp: 192.168.1.117 [192.168.1.117]:5060 ???????????? tcp: 127.0.0.1 [127.0.0.1]:5060 ???????????? tcp: 192.168.1.117 [192.168.1.117]:5060 Aliases: ???????????? tcp: localhost:5060 ???????????? tcp: localhost.localdomain:5060 ???????????? udp: localhost:5060 ???????????? udp: localhost.localdomain:5060 [root@localhost serweb-0.9.6]# /usr/local/sbin/sercmd ERROR: connect_unix_sock: connect(/tmp/ser_ctl): No such file or directory [2] can someone tell me why am i getting that error. Roman ________________________________________________________________________ AOL now offers free email to everyone. Find out more about what's free from AOL at AOL.com.

2 1

[Serusers] Problem with hung up.
by dciepiel＠orange.pl 06 Aug '07

06 Aug '07

Hi, I have a problem with my SER. When the call is established an am trying to disconnect the call, the second side doesn`t discoonect. The call is still established. Does anybody know what is going on? Thanx. cheers, Daniel

2 1

[Serusers] SER query...
by Piyush.Bansal＠relianceada.com 06 Aug '07

06 Aug '07

hi all, Currently i was trying to test SER under load conditions. At the load of just 10 simultaneous calls, the following error occured. "ERROR: forward_reply: no 2nd via found in reply" Note: Sometimes the error occured after 100-110 call and sometime it goes upto 1000 calls.. Can anybody help me in this issue.... regds, Piyush Bansal

2 1

Re: [Serusers] Problem with TM initialialization
by Fernando Schmitt 06 Aug '07

06 Aug '07

Guys, Its stange, because I guess this module doesnt depend on anyone else. Am I wrong? Im trying to run it in a i686 kernel. May this be the problem? version: ser 0.8.11-r1 (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 1.1 2006/11/22 15:56:27 malmeida Exp $ main.c compiled on 06:45:02 Aug 3 2007 with gcc-3.3 3.3 nocserver:/tools/ciscoenvcontrol# uname -a Linux nocserver 2.6.15-23-686 #1 SMP PREEMPT Tue May 23 14:03:07 UTC 2006 i686 GNU/Linux The ser.cfg file is the default, as you can see bellow: # ----------- global configuration parameters ------------------------ debug=10 # debug level (cmd line: -dddddddddd) 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" listen=200.231.193.7 # ------------------ 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" loadmodule "/usr/local/lib/ser/modules/textops.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" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- 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 (len_gt( 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 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); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); 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) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; 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(); }; } Thanks in advance! Fernando ________________________________________ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Fernando Schmitt Sent: Friday, August 03, 2007 9:17 PM To: serusers(a)lists.iptel.org Subject: [Serusers] Problem with TM initialialization Guys, Im receiving the following output when running ser: 0(5705) Aug 03 06:58:26 DEBUG: register_fifo_cmd: new command (sl_stats) registered 0(5705) Aug 03 06:58:26 DEBUG: MD5 calculated: 103d8e7e9fc1a3538c1143a96b22bab4 0(5705) Aug 03 06:58:26 DEBUG: init_mod: tm 0(5705) Aug 03 06:58:26 TM - initializing... 0(5705) Aug 03 06:58:26 Call-ID initialization: '79dff3eb' 0(5705) Aug 03 06:58:26 DEBUG: register_fifo_cmd: new command (t_uac_dlg) registered 0(5705) Aug 03 06:58:26 DEBUG: register_fifo_cmd: new command (t_hash) registered 0(5705) Aug 03 06:58:26 DEBUG: lock_initialize: lock initialization started 0(5705) Aug 03 06:58:26 init_mod(): Error while initializing module tm ERROR: error while initializing modules Do you know why? Thanks!

2 2

Re: [Serusers] need help with media proxy
by samuel 06 Aug '07

06 Aug '07

2007/8/6, Kirk McCalla <kirk.mccalla(a)gmail.com>: > > Hi Samuel this is what I did yesterday. > I loaded media proxy on my ser server as well as my remote server. > I commented the dispatcher part in the mediaproxy remote server ini file > and I commented mediaproxy in the ini file that is loaded on the proxy > server. > > I configiured the ini file in ser to look like this. > > ;start = yes > ;socket = /var/run/proxydispatcher.sock > ;listen=XX.XX.XX.XX PUBLIC IP ADDRESS > ;group = ser > ;defaultProxy =domain://domain.net why do you have ; at the starting of these lines?? Try to remove and see if mediaproxydispather starts #start = yes > #socket = /var/run/mediaproxy.sock > #group = openser > #listen = None > #allow = None > #proxyIP = 10.0.0.1 > #portRange = 60000:65000 > #TOS = 0xb8 > #idleTimeout = 60 > #holdTimeout = 3600 > #forceClose = 0 > > My DNS service records are set up to have > > _mediaproxy._UDP.domain.net 0 0 server.domain.com I would use tcp as it is stated in the mediaproxy README file, it would probably won't work if you setup DNS to redirect traffic to UDP transport... I still cant connect behind a nat > > Am I supposed to see a dispatcher.sock file when media proxy is started on > the ser server. > > Let me know if you can help please. > > Thanks. > > > > On 8/6/07, samuel <samu60(a)gmail.com> wrote: > > > > There's an init script in the mediaproxy sources (under boot directory) > > which would launch mediaproxy dispatcher as stated in the config file. > > Case you configure mediaproxy dispatcher to use domain names, be sure > > DNS is properly configured. > > > > Hope it helps, > > Samuel > > > > 2007/8/5, Kirk McCalla <kirk.mccalla(a)gmail.com >: > > > > > > One more thing how would I start the dispatcher. I am not sur that I > > > have it started. I did a ps -ef|grep proxyldispatcher and ther ewas no Pid > > > file. > > > > > > On 8/4/07, Kirk McCalla <kirk.mccalla(a)gmail.com> wrote: > > > > > > > > Hi I am trying to get media proxy to work with sip express router. > > > > > > > > I am using a pstn configuration that seems to work fine when I am > > > > calling inside my network. > > > > Now when I connect on e UA to an outside network and try to call > > > > the inside network I get dead air. > > > > > > > > this is the error that I get > > > > 6(21198) error: mediaproxy/sendMediaproxyCommand(): can't connect > > > > to MediaProxy > > > > > > > > I have downloaded media proxy and set it up on ser as wekk as my > > > > proxy server. > > > > I have commented out the mediaproxy configs whenmedia server is > > > > runninmg on ser and on the media server box I have commented out the > > > > dispatched infrmation. > > > > > > > > I am using DSL so I have configiured them to listen on my public Ip > > > > address. > > > > > > > > I have followed the install instructions and still cant get it to > > > > work > > > > > > > > does any one have a working example of what the mediaproxy.ini file > > > > should look like. > > > > > > > > ANd shed some light on where I mat be going wrong. > > > > > > > > Thanks > > > > > > > > > > > > > _______________________________________________ > > > Serusers mailing list > > > Serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > >

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[Serusers] bugtracker down?
by Cesc Santa 06 Aug '07

06 Aug '07

Hi, I was trying to get to the bug tracker (from www.iptel.org -> SER ... bug tracker) ... but I get a "502 bad gateway" ... is it just me and my shitty work internet connection or it is down? or what is the correct link? Cesc

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