- sr-users - kamailio.org

[Serusers] Call Forwarding No Answer
by Parminder.Nayyar＠relianceada.com 18 Aug '07

18 Aug '07

Hello I am trying to do call forwarding on no answer. I have configured 3 phones A Party is 1234 B PArty is 1122 and its call is forwarded to C Party ie 4321 on no answer situation. The SER console is printing the following 1(9070) ERROR: t_forward_nonack: no branched for forwarding 1(9070) ERROR: w_t_relay (failure mode): forwarding failed and my config file is attached ... Please help me in finding the error... i need it urgently.. 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] number mapping
by Steffen Waldmann 17 Aug '07

17 Aug '07

hi there, another question: how can i 'map' numbers in SER e.g. my sip username is client1 if i dial 10 it want to get connected to client1 kind regards, steffen

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[OpenSER-Users] Openser with Oracle
by Andre Luiz Martins Rodrigues 17 Aug '07

17 Aug '07

Hello all, I need one information. It's possible to use Openser with Oracle?? For this, is necessary use the UnixODBC?? Att Andre Luiz Martins Rodrigues <mailto:andre.martins@govoip.com.br> mailto:andre.martins@govoip.com.br Fone: +55 (11) 4919-0180 Cel: +55 (34) 9195-3444

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[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

3 4

[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

2 2

[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

2 2

[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 > > > > > > > > >

1 0

[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

2 2

[OpenSER-Users] About openseradmin- Gem install mysql?
by zze-WU Yin RD-MAPS-LAN 06 Aug '07

06 Aug '07

Hi, everybody I have a problem for installation of openseradmin. It must do: gem install mysql [root@opensersip openseradmin-0.3]# gem install mysql Bulk updating Gem source index for: http://gems.rubyforge.org Select which gem to install for your platform (i686-linux) 1. mysql 2.7.3 (mswin32) 2. mysql 2.7.1 (mswin32) 3. mysql 2.7 (ruby) 4. mysql 2.6 (ruby) 5. Skip this gem 6. Cancel installation > 1 Successfully installed mysql-2.7.3-mswin32 Installing ri documentation for mysql-2.7.3-mswin32... Installing RDoc documentation for mysql-2.7.3-mswin32... While generating documentation for mysql-2.7.3-mswin32 ... MESSAGE: Unhandled special: Special: type=17, text="" ... RDOC args: --op /usr/local/lib/ruby/gems/1.8/doc/mysql-2.7.3-mswin32/rdoc --exclude ext --main README --quiet ext README docs/README.html (continuing with the rest of the installation) [root@opensersip openseradmin-0.3]# But I don't know which mysql I have to choose and I choose 1, but I don't understand what means of this: continuing with the rest of the installation Do anybody know? Thank you very much!!! Yin

2 1

[Serusers] need help with media proxy
by Kirk McCalla 06 Aug '07

06 Aug '07

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

2 2

[OpenSER-Users] Problem with openseradmin:login error? Help!!
by zze-WU Yin RD-MAPS-LAN 06 Aug '07

06 Aug '07

Hi everybody, I used a openseradmin who is based on rubygem (i suppose ), so when I start my server of this openseradmin in the http://localhost:3000, I reciceved some messages like this: In the web page: SocketError in LoginController#login getaddrinfo: Name or service not known RAILS_ROOT: script/../config/.. Application Trace | Framework Trace | Full Trace /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ vendor/mysql.rb:111:in `initialize' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ vendor/mysql.rb:111:in `new' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ vendor/mysql.rb:111:in `real_connect' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ connection_adapters/mysql_adapter.rb:389:in `connect' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ connection_adapters/mysql_adapter.rb:152:in `initialize' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ connection_adapters/mysql_adapter.rb:82:in `new' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ connection_adapters/mysql_adapter.rb:82:in `mysql_connection' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ connection_adapters/abstract/connection_specification.rb:262:in `send' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ connection_adapters/abstract/connection_specification.rb:262:in `connection_without_query_cache=' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ query_cache.rb:54:in `connection=' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ connection_adapters/abstract/connection_specification.rb:230:in `retrieve_connection' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ connection_adapters/abstract/connection_specification.rb:78:in `connection' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:1456:in `quote_bound_value' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:1427:in `replace_bind_variables' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:1427:in `gsub' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:1427:in `replace_bind_variables' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:1416:in `sanitize_sql_array' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:1387:in `sanitize_sql' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:1163:in `add_conditions!' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:1096:in `construct_finder_sql' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:997:in `find_every' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:991:in `find_initial' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:417:in `find' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ deprecated_finders.rb:23:in `find_first_without_deprecation' /usr/local/lib/ruby/gems/1.8/gems/activesupport-1.4.2/lib/active_support /deprecation.rb:94:in `find_first' app/models/user.rb:7:in `authenticate' app/controllers/login_controller.rb:11:in `login' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ vendor/mysql.rb:111:in `initialize' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ vendor/mysql.rb:111:in `new' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ vendor/mysql.rb:111:in `real_connect' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ connection_adapters/mysql_adapter.rb:389:in `connect' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ connection_adapters/mysql_adapter.rb:152:in `initialize' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ connection_adapters/mysql_adapter.rb:82:in `new' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ connection_adapters/mysql_adapter.rb:82:in `mysql_connection' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ connection_adapters/abstract/connection_specification.rb:262:in `send' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ connection_adapters/abstract/connection_specification.rb:262:in `connection_without_query_cache=' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ query_cache.rb:54:in `connection=' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ connection_adapters/abstract/connection_specification.rb:230:in `retrieve_connection' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ connection_adapters/abstract/connection_specification.rb:78:in `connection' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:1456:in `quote_bound_value' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:1427:in `replace_bind_variables' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:1427:in `gsub' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:1427:in `replace_bind_variables' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:1416:in `sanitize_sql_array' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:1387:in `sanitize_sql' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:1163:in `add_conditions!' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:1096:in `construct_finder_sql' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:997:in `find_every' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:991:in `find_initial' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:417:in `find' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ deprecated_finders.rb:23:in `find_first_without_deprecation' /usr/local/lib/ruby/gems/1.8/gems/activesupport-1.4.2/lib/active_support /deprecation.rb:94:in `find_first' /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.13.3/lib/action_controlle r/base.rb:1095:in `send' /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.13.3/lib/action_controlle r/base.rb:1095:in `perform_action_without_filters' /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.13.3/lib/action_controlle r/filters.rb:632:in `call_filter' /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.13.3/lib/action_controlle r/filters.rb:634:in `call_filter' /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.13.3/lib/action_controlle r/filters.rb:619:in `perform_action_without_benchmark' /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.13.3/lib/action_controlle r/benchmarking.rb:66:in `perform_action_without_rescue' /usr/local/lib/ruby/1.8/benchmark.rb:293:in `measure' /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.13.3/lib/action_controlle r/benchmarking.rb:66:in `perform_action_without_rescue' /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.13.3/lib/action_controlle r/rescue.rb:83:in `perform_action' /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.13.3/lib/action_controlle r/base.rb:430:in `send' /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.13.3/lib/action_controlle r/base.rb:430:in `process_without_filters' /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.13.3/lib/action_controlle r/filters.rb:624:in `process_without_session_management_support' /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.13.3/lib/action_controlle r/session_management.rb:114:in `process' /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.13.3/lib/action_controlle r/base.rb:330:in `process' /usr/local/lib/ruby/gems/1.8/gems/rails-1.2.3/lib/dispatcher.rb:41:in `dispatch' /usr/local/lib/ruby/gems/1.8/gems/rails-1.2.3/lib/webrick_server.rb:113: in `handle_dispatch' /usr/local/lib/ruby/gems/1.8/gems/rails-1.2.3/lib/webrick_server.rb:79:i n `service' /usr/local/lib/ruby/1.8/webrick/httpserver.rb:104:in `service' /usr/local/lib/ruby/1.8/webrick/httpserver.rb:65:in `run' /usr/local/lib/ruby/1.8/webrick/server.rb:173:in `start_thread' /usr/local/lib/ruby/1.8/webrick/server.rb:162:in `start' /usr/local/lib/ruby/1.8/webrick/server.rb:162:in `start_thread' /usr/local/lib/ruby/1.8/webrick/server.rb:95:in `start' /usr/local/lib/ruby/1.8/webrick/server.rb:92:in `each' /usr/local/lib/ruby/1.8/webrick/server.rb:92:in `start' /usr/local/lib/ruby/1.8/webrick/server.rb:23:in `start' /usr/local/lib/ruby/1.8/webrick/server.rb:82:in `start' /usr/local/lib/ruby/gems/1.8/gems/rails-1.2.3/lib/webrick_server.rb:63:i n `dispatch' /usr/local/lib/ruby/gems/1.8/gems/rails-1.2.3/lib/commands/servers/webri ck.rb:59 /usr/local/lib/ruby/site_ruby/1.8/rubygems/custom_require.rb:27:in `gem_original_require' /usr/local/lib/ruby/site_ruby/1.8/rubygems/custom_require.rb:27:in `require' /usr/local/lib/ruby/gems/1.8/gems/activesupport-1.4.2/lib/active_support /dependencies.rb:495:in `require' /usr/local/lib/ruby/gems/1.8/gems/activesupport-1.4.2/lib/active_support /dependencies.rb:342:in `new_constants_in' /usr/local/lib/ruby/gems/1.8/gems/activesupport-1.4.2/lib/active_support /dependencies.rb:495:in `require' /usr/local/lib/ruby/gems/1.8/gems/rails-1.2.3/lib/commands/server.rb:39 /usr/local/lib/ruby/site_ruby/1.8/rubygems/custom_require.rb:27:in `gem_original_require' /usr/local/lib/ruby/site_ruby/1.8/rubygems/custom_require.rb:27:in `require' /usr/local/lib/ruby/gems/1.8/gems/activesupport-1.4.2/lib/active_support /dependencies.rb:495:in `require' /usr/local/lib/ruby/gems/1.8/gems/activesupport-1.4.2/lib/active_support /dependencies.rb:342:in `new_constants_in' /usr/local/lib/ruby/gems/1.8/gems/activesupport-1.4.2/lib/active_support /dependencies.rb:495:in `require' script/server:3 /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ vendor/mysql.rb:111:in `initialize' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ vendor/mysql.rb:111:in `new' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ vendor/mysql.rb:111:in `real_connect' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ connection_adapters/mysql_adapter.rb:389:in `connect' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ connection_adapters/mysql_adapter.rb:152:in `initialize' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ connection_adapters/mysql_adapter.rb:82:in `new' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ connection_adapters/mysql_adapter.rb:82:in `mysql_connection' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ connection_adapters/abstract/connection_specification.rb:262:in `send' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ connection_adapters/abstract/connection_specification.rb:262:in `connection_without_query_cache=' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ query_cache.rb:54:in `connection=' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ connection_adapters/abstract/connection_specification.rb:230:in `retrieve_connection' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ connection_adapters/abstract/connection_specification.rb:78:in `connection' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:1456:in `quote_bound_value' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:1427:in `replace_bind_variables' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:1427:in `gsub' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:1427:in `replace_bind_variables' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:1416:in `sanitize_sql_array' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:1387:in `sanitize_sql' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:1163:in `add_conditions!' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:1096:in `construct_finder_sql' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:997:in `find_every' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:991:in `find_initial' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ base.rb:417:in `find' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.15.3/lib/active_record/ deprecated_finders.rb:23:in `find_first_without_deprecation' /usr/local/lib/ruby/gems/1.8/gems/activesupport-1.4.2/lib/active_support /deprecation.rb:94:in `find_first' app/models/user.rb:7:in `authenticate' app/controllers/login_controller.rb:11:in `login' /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.13.3/lib/action_controlle r/base.rb:1095:in `send' /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.13.3/lib/action_controlle r/base.rb:1095:in `perform_action_without_filters' /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.13.3/lib/action_controlle r/filters.rb:632:in `call_filter' /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.13.3/lib/action_controlle r/filters.rb:634:in `call_filter' /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.13.3/lib/action_controlle r/filters.rb:619:in `perform_action_without_benchmark' /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.13.3/lib/action_controlle r/benchmarking.rb:66:in `perform_action_without_rescue' /usr/local/lib/ruby/1.8/benchmark.rb:293:in `measure' /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.13.3/lib/action_controlle r/benchmarking.rb:66:in `perform_action_without_rescue' /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.13.3/lib/action_controlle r/rescue.rb:83:in `perform_action' /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.13.3/lib/action_controlle r/base.rb:430:in `send' /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.13.3/lib/action_controlle r/base.rb:430:in `process_without_filters' /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.13.3/lib/action_controlle r/filters.rb:624:in `process_without_session_management_support' /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.13.3/lib/action_controlle r/session_management.rb:114:in `process' /usr/local/lib/ruby/gems/1.8/gems/actionpack-1.13.3/lib/action_controlle r/base.rb:330:in `process' /usr/local/lib/ruby/gems/1.8/gems/rails-1.2.3/lib/dispatcher.rb:41:in `dispatch' /usr/local/lib/ruby/gems/1.8/gems/rails-1.2.3/lib/webrick_server.rb:113: in `handle_dispatch' /usr/local/lib/ruby/gems/1.8/gems/rails-1.2.3/lib/webrick_server.rb:79:i n `service' /usr/local/lib/ruby/1.8/webrick/httpserver.rb:104:in `service' /usr/local/lib/ruby/1.8/webrick/httpserver.rb:65:in `run' /usr/local/lib/ruby/1.8/webrick/server.rb:173:in `start_thread' /usr/local/lib/ruby/1.8/webrick/server.rb:162:in `start' /usr/local/lib/ruby/1.8/webrick/server.rb:162:in `start_thread' /usr/local/lib/ruby/1.8/webrick/server.rb:95:in `start' /usr/local/lib/ruby/1.8/webrick/server.rb:92:in `each' /usr/local/lib/ruby/1.8/webrick/server.rb:92:in `start' /usr/local/lib/ruby/1.8/webrick/server.rb:23:in `start' /usr/local/lib/ruby/1.8/webrick/server.rb:82:in `start' /usr/local/lib/ruby/gems/1.8/gems/rails-1.2.3/lib/webrick_server.rb:63:i n `dispatch' /usr/local/lib/ruby/gems/1.8/gems/rails-1.2.3/lib/commands/servers/webri ck.rb:59 /usr/local/lib/ruby/site_ruby/1.8/rubygems/custom_require.rb:27:in `gem_original_require' /usr/local/lib/ruby/site_ruby/1.8/rubygems/custom_require.rb:27:in `require' /usr/local/lib/ruby/gems/1.8/gems/activesupport-1.4.2/lib/active_support /dependencies.rb:495:in `require' /usr/local/lib/ruby/gems/1.8/gems/activesupport-1.4.2/lib/active_support /dependencies.rb:342:in `new_constants_in' /usr/local/lib/ruby/gems/1.8/gems/activesupport-1.4.2/lib/active_support /dependencies.rb:495:in `require' /usr/local/lib/ruby/gems/1.8/gems/rails-1.2.3/lib/commands/server.rb:39 /usr/local/lib/ruby/site_ruby/1.8/rubygems/custom_require.rb:27:in `gem_original_require' /usr/local/lib/ruby/site_ruby/1.8/rubygems/custom_require.rb:27:in `require' /usr/local/lib/ruby/gems/1.8/gems/activesupport-1.4.2/lib/active_support /dependencies.rb:495:in `require' /usr/local/lib/ruby/gems/1.8/gems/activesupport-1.4.2/lib/active_support /dependencies.rb:342:in `new_constants_in' /usr/local/lib/ruby/gems/1.8/gems/activesupport-1.4.2/lib/active_support /dependencies.rb:495:in `require' script/server:3 Request Parameters: {"user_login"=>"openser", "user_password"=>"openserrw", "login"=>"Login >"} Show session dump --- return-to: / flash: !map:ActionController::Flash::FlashHash {} Response Headers: {"cookie"=>[], "Cache-Control"=>"no-cache"} Do anybody know what's meanning of this log? And how can I resolve it? Thank you very much!!!! Yin

2 1

[OpenSER-Users] loadmodule "perl.so"
by Richard Timsit 06 Aug '07

06 Aug '07

Hey Everyone, when i try to load module perl.so, i get this message : ERROR: load_module: could not open module </usr/local/lib/openser/modules/perl.so>: /usr/local/lib/openser/modules/perl.so: undefined symbol: boot_OpenSER In /usr/local/lib/openser/modules/ i have not only perl.so but also the lib/perl dir. openser-1.2.0-tls or openser-1.2.1-tls Any idea ? Thanks, -- Richard Timsit <Richard.Timsit(a)epfl.ch> EPFL

3 5

[Serusers] dealing with parallel forking ...
by Cesc Santa 06 Aug '07

06 Aug '07

Hi, Question is: how do you deal with the multiple 200 OK responses that parallel forking in ser (using append_branches and then t_forward_nonack) deliver? I am doing some tests and it is not really a problem ... but maybe someone has a better idea. In my configuration, the first 200 OK received is forwarded to the caller and the whole SIP session setup (caller + 1st callee). Next 200 OKs are also delivered to the caller, who happily ignores them ... it is the task of the 2nd (and 3rd, ... ) callee to at a certain point give up and send a BYE, to which the caller replies with 481 no call leg ... It all works ... but, I wonder if ser could send CANCELs after receiving the 1st 200 OK ... if yes, how? :) I have a route_reply setup ... but only the first 200 OK gets shown ... next 200 OKs are not matched to any transaction and thus "silently" forwarded ... (via) Cesc

3 11

[OpenSER-Users] [OT] ngrep creates a binary file
by Iñaki Baz Castillo 06 Aug '07

06 Aug '07

Hi, I use ngrep command for debuggin but it creates a binary file instead of text file ~# ngrep -P '' -W byline "" udp port 5060 > FILE.LOG ~# file FILE.LOG FILE.LOG: data but I want to get a text file or I have some problems in file editors and so. I haven't saw option in ngrep for my issue, does somebody know how ngrep to generate text files isntead of binary files? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

2 2

[Serusers] announcement playing using SER
by Piyush.Bansal＠relianceada.com 06 Aug '07

06 Aug '07

Hi all, Is there any means thru which i can play announcement / tones from SER???? Plz revert back ASAP...

1 0

[Serusers] [t_uac_dlg] transactionid problem in ACK for 2xx response
by KevinKuei 06 Aug '07

06 Aug '07

Dear all, I found a problem in using t_uac_dlg to send INVITE via FIFO. When it got a 200 OK response, it sends ACK automaticlly. I saw the Via branch parameter value in the ACK is the same with the value in INVITE. But based on RFC3261 sec.13.2.2, and sec.8.1.1.7, the branch parameter value in ACK should not be the same for 2xx response. Is there any patch code?? -- Kevin Kuei ---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

1 0

[OpenSER-Users] problem with sipwise config
by Alexander J. Perovich 05 Aug '07

05 Aug '07

Hello, I am having issues with the openser.cfg from sipwise. The CFC does not forward the calls when set in the database. Also, I am using the full uri as the UUID for example UUID can be 4163334444(a)proxy.domain.net. I have modified the config for radius authentication as well. Here is what I am trying to do: 1. Radius Authentication on register and PSTN calling 2. CFC based on user preferences in database 3. If user offline forward the asterisk vmail server 4. If user cancels call it will forward to vmail server Here is the config I currently have: ######################################################################## # This configuration is autogenerated by sip:wizard # (http://www.sipwise.com/wizard) on Tue Jun 26 22:47:33 +0200 2007 # for OpenSER 1.2 # # Copyright (C) 2007 Sipwise (support(a)sipwise.com) ######################################################################## ######################################################################## # By obtaining, using, and/or copying this configuration and/or its # associated documentation, you agree that you have read, understood, # and will comply with the Terms of Usage provided at # http://www.sipwise.com/news/?page_id=6 as well as the following # additions: # # Permission to use, copy, modify, and distribute this configuration and # its associated documentation for any purpose and without fee is hereby # granted, provided that the above copyright notice appears in all # copies, and that both that copyright notice and this permission notice # appear in supporting documentation, and that the name of Sipwise or # the author will not be used in advertising or publicity pertaining to # distribution of the configuration without specific, written prior # permission. ######################################################################## ######################################################################## # Before using this configuration, read the following prerequisites in # order to gain the designated functionallity: # # base: # You have to insert all locally served domains (i.e. # "openserctl domain add your.domain.com"). # # nat-mediaproxy: # You have to install mediaproxy # (http://mediaproxy.ag-projects.com/) for relaying RTP traffic. # # usr-preferences: # This feature relies on UUID-based provisioning. Thus, you have # to add the uuid-column to the subscriber table ("alter table # subscriber add column uuid varchar(64);") and populate it with a # UUID (unique user identifier) per user, for example an # auto-incremented id. # # offnet-incoming-sip: # You have to populate the "trusted"-table with rules for # allowed peering hosts (i.e. src_ip="1.2.3.4", proto="udp", # from_pattern="^sip:.*@domain.of.peering.host$", tag="1234"). If the # feature "usr-preferences" is selected, the tag-value is used as # caller-uuid for such calls. # # offnet-pstn: # You have to add a routing entry for lcr (i.e. "openserctl lcr # addroute '' '' 1 1"). Additionally, you have to add your gateways # (i.e. "openserctl lcr addgw my-test-gw 1.2.3.4 5060 sip udp 1"). # # cfu: # You have to provision the call-forward-unconditional as full # SIP URI (AVP cfu as type 0) for each user in the usr_preferences # table (i.e. uuid='1234', username='', domain='', attribute='cfu', # type=0, value='sip:foo@otherdomain.com'). Forwards to another user # in the same domain or to other domains are possible. # # cfc: # You have to provision the call-forward-conditional as full SIP # URI (AVP cfc as type 0) for each user in the usr_preferences table # (i.e. uuid='1234', username='', domain='', attribute='cfc', type=0, # value='sip:foo@otherdomain.com'). Forwards to another user in the # same domain or to other domains are possible. # # user-aliases: # You have to add aliases for your users (i.e. "openserctl alias # add 01234567 sip:bob@yourdomain.com" for usrloc-based aliases or # make entries into "dbaliases" table for db-based aliases) # # cli: # You have to provision the CLI as full SIP URI (AVP cli as type # 0) for each user in the usr_preferences table (i.e. uuid='1234', # username='', domain='', attribute='cli', type=0, # value='sip:01234567@yourdomain.com'). # # clir: # You have to provision '1' to enable CLIR and '0' to disable it # (AVP clir as type 1) for each user in the usr_preferences table # (i.e. uuid='1234', username='', domain='', attribute='clir', # type=1, value='1'). # # acc-db: # You have to add the columns "src_leg" and "dst_leg" to your # acc-table ("alter table acc add column src_leg varchar(128); alter # table acc add column dst_leg varchar(128);"). # ######################################################################## ######################################################################## # Configuration 'sip:wizard - Tue Jun 26 22:47:33 +0200 2007' ######################################################################## listen = udp::x.x.x.x:5060 mpath = "/usr/local/lib/openser/modules" children = 8 debug = 3 fork = yes group = "openser" user = "openser" disable_tcp = no log_facility = LOG_DAEMON log_stderror = yes tcp_children = 4 mhomed = no server_signature = yes sock_group = "openser" sock_mode = 0600 sock_user = "openser" unix_sock = "/tmp/openser.sock" unix_sock_children = 1 reply_to_via = no sip_warning = no check_via = no dns = no rev_dns = no disable_core_dump = no dns_try_ipv6 = yes dns_use_search_list = yes loadmodule "usrloc.so" modparam("usrloc", "user_column", "username") modparam("usrloc", "domain_column", "domain") modparam("usrloc", "contact_column", "contact") modparam("usrloc", "expires_column", "expires") modparam("usrloc", "q_column", "q") modparam("usrloc", "callid_column", "callid") modparam("usrloc", "cseq_column", "cseq") modparam("usrloc", "methods_column", "methods") modparam("usrloc", "flags_column", "flags") modparam("usrloc", "user_agent_column", "user_agent") modparam("usrloc", "received_column", "received") modparam("usrloc", "socket_column", "socket") modparam("usrloc", "use_domain", 0) modparam("usrloc", "desc_time_order", 0) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "db_url", "mysql://openser:xxxx@x.x.x.x/openser") modparam("usrloc", "db_mode", 1) modparam("usrloc", "matching_mode", 0) modparam("usrloc", "cseq_delay", 20) modparam("usrloc", "nat_bflag", 6) loadmodule "registrar.so" modparam("registrar", "default_expires", 3600) modparam("registrar", "min_expires", 60) modparam("registrar", "max_expires", 0) modparam("registrar", "default_q", 0) modparam("registrar", "append_branches", 1) modparam("registrar", "case_sensitive", 0) modparam("registrar", "received_param", "received") modparam("registrar", "max_contacts", 0) modparam("registrar", "retry_after", 0) modparam("registrar", "method_filtering", 0) modparam("registrar", "path_mode", 2) modparam("registrar", "path_use_received", 0) modparam("registrar", "received_avp", "$avp(i:801)") loadmodule "rr.so" modparam("rr", "enable_full_lr", 0) modparam("rr", "append_fromtag", 1) modparam("rr", "enable_double_rr", 1) modparam("rr", "add_username", 0) loadmodule "tm.so" modparam("tm", "fr_timer", 30) modparam("tm", "fr_inv_timer", 120) modparam("tm", "wt_timer", 5) modparam("tm", "delete_timer", 2) modparam("tm", "noisy_ctimer", 0) modparam("tm", "ruri_matching", 1) modparam("tm", "via1_matching", 1) modparam("tm", "unix_tx_timeout", 2) modparam("tm", "restart_fr_on_each_reply", 1) modparam("tm", "pass_provisional_replies", 0) loadmodule "xlog.so" modparam("xlog", "buf_size", 4096) modparam("xlog", "force_color", 0) loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("mi_fifo", "fifo_mode", 0660) modparam("mi_fifo", "fifo_group", "openser") modparam("mi_fifo", "fifo_user", "openser") modparam("mi_fifo", "reply_dir", "/tmp/") modparam("mi_fifo", "reply_indent", "\t") loadmodule "domain.so" modparam("domain", "db_url", "mysql://openser:xxxx@x.x.x.x/openser") modparam("domain", "db_mode", 1) modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 0) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "received_avp", "$avp(i:801)") loadmodule "textops.so" loadmodule "mediaproxy.so" modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("mediaproxy", "sip_asymmetrics", "/etc/openser/sip-asymmetric-clients") modparam("mediaproxy", "rtp_asymmetrics", "/etc/openser/rtp-asymmetric-clients") modparam("mediaproxy", "natping_interval", 60) loadmodule "uri.so" loadmodule "sl.so" modparam("sl", "enable_stats", 1) loadmodule "maxfwd.so" modparam("maxfwd", "max_limit", 256) loadmodule "mysql.so" modparam("mysql", "ping_interval", 300) modparam("mysql", "auto_reconnect", 1) loadmodule "auth.so" modparam("auth", "nonce_expire", 300) modparam("auth", "rpid_suffix", ";party=calling;id-type=subscriber;screen=yes") modparam("auth", "rpid_avp", "$avp(s:rpid)") loadmodule "auth_db.so" modparam("auth_db", "db_url", "mysql://openser:xxxx@x.x.x.x/openser") modparam("auth_db", "user_column", "username") modparam("auth_db", "domain_column", "domain") modparam("auth_db", "password_column", "password") modparam("auth_db", "password_column_2", "ha1b") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "use_domain", 0) modparam("auth_db", "load_credentials", "$avp(s:caller_uuid)=uuid") loadmodule "uri_db.so" modparam("uri_db", "db_url", "mysql://openser:xxxx@x.x.x.x/openser") modparam("uri_db", "uri_table", "uri") modparam("uri_db", "uri_user_column", "username") modparam("uri_db", "uri_domain_column", "domain") modparam("uri_db", "uri_uriuser_column", "uri_user") modparam("uri_db", "subscriber_table", "subscriber") modparam("uri_db", "subscriber_user_column", "username") modparam("uri_db", "subscriber_domain_column", "domain") modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "use_domain", 0) loadmodule "avpops.so" modparam("avpops", "db_url", "mysql://openser:xxxx@x.x.x.x/openser") modparam("avpops", "avp_table", "usr_preferences") modparam("avpops", "use_domain", 0) modparam("avpops", "uuid_column", "uuid") modparam("avpops", "username_column", "username") modparam("avpops", "domain_column", "domain") modparam("avpops", "attribute_column", "attribute") modparam("avpops", "value_column", "value") modparam("avpops", "type_column", "type") loadmodule "permissions.so" modparam("permissions", "default_allow_file", "permissions.allow") modparam("permissions", "default_deny_file", "permissions.deny") modparam("permissions", "check_all_branches", 1) modparam("permissions", "allow_suffix", ".allow") modparam("permissions", "deny_suffix", ".deny") modparam("permissions", "db_url", "mysql://openser:xxxx@x.x.x.x/openser") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "source_col", "src_ip") modparam("permissions", "proto_col", "proto") modparam("permissions", "from_col", "from_pattern") modparam("permissions", "tag_col", "tag") modparam("permissions", "peer_tag_avp", "s:peer_uuid") loadmodule "lcr.so" modparam("lcr", "db_url", "mysql://openser:xxxx@x.x.x.x/openser") modparam("lcr", "gw_table", "gw") modparam("lcr", "gw_name_column", "gw_name") modparam("lcr", "ip_addr_column", "ip_addr") modparam("lcr", "port_column", "port") modparam("lcr", "uri_scheme_column", "uri_scheme") modparam("lcr", "transport_column", "transport") modparam("lcr", "grp_id_column", "grp_id") modparam("lcr", "lcr_table", "lcr") modparam("lcr", "strip_column", "strip") modparam("lcr", "prefix_column", "prefix") modparam("lcr", "from_uri_column", "from_uri") modparam("lcr", "priority_column", "priority") modparam("lcr", "gw_uri_avp", "1400") modparam("lcr", "ruri_user_avp", "1402") modparam("lcr", "contact_avp", "1401") modparam("lcr", "fr_inv_timer_avp", "s:fr_inv_timer_avp") modparam("lcr", "fr_inv_timer", 90) modparam("lcr", "fr_inv_timer_next", 30) modparam("lcr", "rpid_avp", "s:rpid") loadmodule "alias_db.so" modparam("alias_db", "db_url", "mysql://openser:xxxx@x.x.x.x/openser") modparam("alias_db", "user_column", "username") modparam("alias_db", "domain_column", "domain") modparam("alias_db", "alias_user_column", "alias_username") modparam("alias_db", "alias_domain_column", "alias_domain") modparam("alias_db", "use_domain", 0) loadmodule "uac.so" modparam("uac", "rr_store_param", "vsf") modparam("uac", "from_restore_mode", "auto") modparam("uac", "from_passwd", "s1p:Wiz4rd!") #loadmodule "acc.so" #modparam("acc", "early_media", 0) #modparam("acc", "failed_transaction_flag", 24) #modparam("acc", "report_ack", 0) #modparam("acc", "report_cancels", 0) #modparam("acc", "log_flag", 0) #modparam("acc", "log_missed_flag", 0) #modparam("acc", "log_level", 2) loadmodule "auth_radius.so" loadmodule "group_radius.so" loadmodule "avp_radius.so" modparam("auth_radius|group_radius|avp_radius", "radius_config","/etc/radiusclient-ng/radiusclient.conf") loadmodule "acc.so" modparam("acc", "failed_transaction_flag", 1) modparam("acc", "report_cancels", 0) modparam("acc", "report_ack", 0) modparam("acc", "early_media", 0) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 1) modparam("acc", "db_flag", 25) modparam("acc", "db_missed_flag", 0) modparam("acc", "db_table_acc", "acc") modparam("acc", "db_table_missed_calls", "missed_calls") modparam("acc", "db_url", "mysql://openser:xxxx@x.x.x.x/openser") modparam("acc", "acc_method_column", "method") modparam("acc", "acc_callid_column", "callid") modparam("acc", "acc_time_column", "time") modparam("acc", "acc_totag_column", "to_tag") modparam("acc", "acc_from_tag_column", "from_tag") modparam("acc", "detect_direction", 1) modparam("acc", "acc_sip_code_column", "sip_code") modparam("acc", "acc_sip_reason_column", "sip_reason") #modparam("acc", "multi_leg_info", "leg_src=$avp(i:901);leg_dst=$avp(i:902)") modparam("acc", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 1) #modparam("acc", "multi_leg_info", "RAD_LEG_SRC=$avp(i:901);RAD_LEG_SRC=$avp(i:902)") modparam("acc", "radius_extra", "Sip-RPid=$avp(s:rpid); \ Source-IP=$si; \ Source-Port=$sp; \ Canonical-URI=$ru; \ Billing-Party=$avp(s:billingparty); \ Divert-Reason=$avp(s:divert_reason); \ X-RTP-Stat=$avp(s:rtp_statistics); \ From-Header=$hdr(from); \ User-Agent=$hdr(user-agent); \ Contact=$hdr(contact); \ Calling-Station-Id=$fu; \ Called-Station-Id=$rU; \ Event=$hdr(event); \ SIP-Proxy-IP=$avp(s:sip_proxy_ip)") # Billing-Party=$avp(s:caller_uuid); \ loadmodule "siptrace.so" modparam("siptrace", "trace_flag", 0) modparam("siptrace", "db_url", "mysql://openser:xxxx@x.x.x.x/openser") modparam("siptrace", "trace_on", 1) modparam("siptrace", "table", "sip_trace") modparam("siptrace", "duplicate_uri", "sip:x.x.x.x:5888") ######################################################################## # Request route 'main' ######################################################################## route[0] { sip_trace(); #ajp sip trace to database xlog("L_INFO", "New request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); force_rport(); if(msg:len > max_len) { xlog("L_INFO", "Message too big - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("513", "Message Too Big"); exit; } if (!mf_process_maxfwd_header("10")) { xlog("L_INFO", "Too many hops - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("483", "Too Many Hops"); exit; } if(!is_method("REGISTER")) { if(nat_uac_test("19")) { record_route(";nat=yes"); } else { record_route(); } } if(is_method("CANCEL") || is_method("BYE")) { setflag(1); end_media_session(); } if(loose_route()) { if(!has_totag()) { xlog("L_INFO", "Initial loose-routing rejected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Initial Loose-Routing Rejected"); exit; } if(nat_uac_test("19") || search("^Route:.*;nat=yes")) { fix_nated_contact(); setbflag(6); } if(is_method("BYE")) { setflag(24); # account failed transactions setflag(25); # account successful transactions } # mark as loose-routed for acc setflag(26); route(12); } if(is_method("REGISTER")) { route(11); } setflag(24); # account failed transactions setflag(25); # account successful transactions if(is_method("INVITE")) { route(13); } if(is_method("CANCEL") || is_method("ACK")) { route(17); } route(18); } ######################################################################## # Request route 'clear-usr-preferences-caller' ######################################################################## route[1] { xlog("L_INFO", "Clear caller preferences - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); avp_delete("$avp(s:caller_cli)/g"); avp_delete("$avp(s:clir)/g"); } ######################################################################## # Request route 'clear-usr-preferences-callee' ######################################################################## route[2] { xlog("L_INFO", "Clear callee preferences - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); avp_delete("$avp(s:cfu)/g"); avp_delete("$avp(s:cfc)/g"); } ######################################################################## # Request route 'usr-preferences-caller' ######################################################################## route[3] { route(1); xlog("L_INFO", "Load caller preferences for uuid '$avp(s:caller_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # load caller avps avp_db_load("$avp(s:caller_uuid)", "*"); avp_copy("$avp(s:cli)", "$avp(s:caller_cli)/d"); if(is_avp_set("$avp(s:clir)/n") && avp_check("$avp(s:clir)", "eq/i:1")) { # mark for anonymization setflag(28); } } ######################################################################## # Request route 'usr-preferences-callee' ######################################################################## route[4] { xlog("L_INFO", "Load callee preferences for uuid '$avp(s:callee_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # load callee avps avp_db_load("$avp(s:callee_uuid)", "*"); if(is_avp_set("$avp(s:cfu)/s")) { xlog("L_INFO", "Call-forward-unconditional to '$avp(s:cfu)' found - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(8); avp_delete("$avp(s:caller_uuid)/g"); avp_copy("$avp(s:callee_uuid)", "$avp(s:caller_uuid)/d"); avp_pushto("$ru", "$avp(s:cfu)"); route(3); route(14); exit; } } ######################################################################## # Request route 'acc-caller' ######################################################################## route[5] { xlog("L_INFO", "Setting acc source-leg for uuid '$avp(s:caller_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); avp_printf("$avp(i:901)", "$avp(s:caller_uuid)|$avp(s:acc_caller_user)|$avp(s:acc_caller_domain)|$ avp(s:acc_state)"); } ######################################################################## # Request route 'acc-callee' ######################################################################## route[6] { xlog("L_INFO", "Setting acc destination-leg for uuid '$avp(s:callee_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); avp_printf("$avp(i:902)", "$avp(s:callee_uuid)|$avp(s:acc_callee_user)|$avp(s:acc_callee_domain)") ; } ######################################################################## # Request route 'acc-failure' ######################################################################## route[7] { xlog("L_INFO", "Accounting failed request for uuid '$avp(s:caller_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(5); route(6); resetflag(24); acc_db_request("404", "acc"); } ######################################################################## # Request route 'cfu-acc' ######################################################################## route[8] { $avp(s:acc_callee_user) = $rU; $avp(s:acc_callee_domain) = $rd; route(5); route(6); avp_delete("$avp(s:acc_caller_user)"); avp_delete("$avp(s:acc_caller_domain)"); avp_copy("$avp(s:acc_callee_user)", "$avp(s:acc_caller_user)"); avp_copy("$avp(s:acc_callee_domain)", "$avp(s:acc_caller_domain)"); avp_delete("$avp(s:acc_state)/g"); $avp(s:acc_state) = "cfu"; } ######################################################################## # Request route 'clir' ######################################################################## route[9] { if(isflagset(28) && !isflagset(27)) { setflag(27); xlog("L_INFO", "Anonymize caller - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); uac_replace_from("Anonymous","sip:anonymous@anonymous.invalid"); if(is_present_hf("Privacy")) { remove_hf("Privacy"); } append_hf("Privacy: id\r\n"); } } ######################################################################## # Request route 'stop-media-proxy' ######################################################################## route[10] { if(isflagset(22)) { end_media_session(); } } ######################################################################## # Request route 'base-route-register' ######################################################################## route[11] { sl_send_reply("100", "Trying"); if(!www_authorize("", "subscriber") && !radius_www_authorize("")) #if(!radius_www_authorize("proxy-ca.thinksimplicity.net")) { xlog("L_INFO", "Register authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); www_challenge("", "0"); exit; } if(!check_to()) { xlog("L_INFO", "Spoofed To-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed To-URI Detected"); exit; } consume_credentials(); if(!search("^Contact:[ ]*\*") && nat_uac_test("19")) { fix_nated_register(); setbflag(6); } if(!save("location")) { xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_reply_error(); exit; } xlog("L_INFO", "Registration successful - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); exit; } ######################################################################## # Request route 'base-outbound' ######################################################################## route[12] { if(is_present_hf("P-Asserted-Identity")) { remove_hf("P-Asserted-Identity"); } if(is_present_hf("Remote-Party-ID")) { remove_hf("Remote-Party-ID"); } if(is_avp_set("$avp(s:caller_cli)/s")) { if(!isflagset(28)) { xlog("L_INFO", "Set caller CLI '$avp(s:caller_cli)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); append_hf("P-Asserted-Identity: <$avp(s:caller_cli)>\r\n"); append_hf("Remote-Party-ID: <$avp(s:caller_cli)>;party=caller;privacy=none;screen=yes\r\n"); } } route(9); if(isbflagset(6)) { if(!isflagset(22) && !search("^Content-Length:[ ]*0")) { setflag(22); use_media_proxy(); } t_on_reply("2"); } else { t_on_reply("1"); } if(!isflagset(21)) { t_on_failure("2"); if(!isflagset(26)) { $avp(s:acc_callee_user) = $rU; $avp(s:acc_callee_domain) = $rd; route(5); route(6); } } if(isflagset(29)) { append_branch(); } if(is_present_hf("Proxy-Authorization")) { consume_credentials(); } xlog("L_INFO", "Request leaving server, D-URI='$du' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # no 100 (we already sent it) and no DNS blacklisting if(!t_relay("0x05")) { sl_reply_error(); if(is_method("INVITE") && isbflagset(6)) { end_media_session(); } } exit; } ######################################################################## # Request route 'base-route-invite' ######################################################################## route[13] { sl_send_reply("100", "Trying"); if(from_gw()) { $avp(s:caller_uuid) = "0"; xlog("L_INFO", "Call from PSTN' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); #setflag(1); setflag(23); } else { if(allow_trusted()) { if(is_avp_set("$avp(s:peer_uuid)/s")) { # use tag-column from trusted-table as uuid for this caller avp_copy("$avp(s:peer_uuid)", "$avp(s:caller_uuid)/d"); } else { # if no uuid is set, use "0" as default uuid $avp(s:caller_uuid) = "0"; } xlog("L_INFO", "Call from trusted peer with uuid '$avp(s:caller_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(!is_domain_local("$rd")) { xlog("L_INFO", "Rejecting peering attempt with non-local request domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Relaying Denied"); exit; } setflag(23); } } #else #{ ########AJP - Radius Authorization #if (!radius_proxy_authorize("")) #{ # xlog("L_INFO", "Route authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # proxy_challenge("", "0"); #} #if(!check_from()) #{ # xlog("L_INFO", "Spoofed To-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # sl_send_reply("403", "Spoofed To-URI Detected"); #} #consume_credentials(); ######### #} $avp(s:acc_caller_user) = $fU; $avp(s:acc_caller_domain) = $fd; $avp(s:acc_state) = "call"; #setflag(1); #$avp(s:sip-proxy) = "x.x.x.x"; #$avp($ru) = "can_uri"; route(3); if(nat_uac_test("19")) { fix_nated_contact(); setbflag(6); } route(14); } ######################################################################## # Request route 'invite-find-callee' ######################################################################## route[14] { if(lookup("aliases")) { xlog("L_INFO", "Callee was aliased - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } route(2); if(!is_domain_local("$rd")) { setflag(20); $avp(s:callee_uuid) = "0"; route(16); } #######ajp - Strip 1 for possible local numbers if (uri=~"^sip:1[0-9]{10}@") { strip(1); } avp_delete("$avp(s:callee_uuid)"); avp_db_query("select uuid from subscriber where username = '$rU'", "$avp(s:callee_uuid)"); if(is_avp_set("$avp(s:callee_uuid)/s")) { xlog("L_INFO", "Callee is local, uuid='$avp(s:callee_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(15); } else { $avp(s:callee_uuid) = "0"; xlog("L_INFO", "Callee is not local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(16); } exit; } ######################################################################## # Request route 'invite-to-internal' ######################################################################## route[15] { route(4); if(!lookup("location")) { xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); $avp(s:acc_callee_user) = $rU; $avp(s:acc_callee_domain) = $rd; route(7); sl_send_reply("404", "User Offline"); } else { xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); avp_delete("$avp(s:caller_uuid)/g"); #ajp - This call is billed to receiving party. avp_copy("$avp(s:callee_uuid)", "$avp(s:caller_uuid)/d"); $avp(s:billingparty) = "inbound"; #avp_delete("$avp(s:caller_uuid)"); #ajp - billed to internal user #avp_db_query("select uuid from subscriber where username = '$tU'", "$avp(s:caller_uuid)"); #ajp #$avp(s:billingparty) = $avp(s:caller_uuid); $avp(s:sip_proxy_ip) = "x.x.x.x"; setflag(1); #accounting route(12); } exit; } ######################################################################## # Request route 'invite-to-external' ######################################################################## route[16] { if(isflagset(20)) { xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(12); exit; } if(!isflagset(23)) { # don't allow calls relaying from PSTN to PSTN, if not explicitely forwarded if(uri =~ "^sip:[0-9]+@") { # only route numeric users to PSTN #ajp-normalize the call if(uri =~ "^sip:[0-9]{10}+@") #ajp- North Americal local numbers prefix with 1 { prefix("1"); } if(uri =~ "^sip:001[0-9]*@") #ajp- Europe dial to North America { strip(2); } if(uri =~ "^sip:00[0-9]*@") #ajp- International Calling from Europe { strip(2); } if(uri =~ "^sip:011[0-9]*@") #ajp- International Calling from North America { strip(3); } #ajp-accounting #$avp(s:acc_caller_uuid) = $rU; #ajp avp_delete("$avp(s:callee_uuid)"); #ajp avp_db_query("select uuid from subscriber where username = '$fU'", "$avp(s:callee_uuid)"); #ajp $avp(s:billingparty) = $avp(s:callee_uuid); $avp(s:sip_proxy_ip) = "x.x.x.x"; #ajp accounting if(!load_gws()) { xlog("L_ERR", "Error loading PSTN gateways - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } if(!next_gw()) { xlog("L_ERR", "No PSTN gateways available - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } setflag(21); setflag(1); t_on_failure("1"); route(12); } } xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(7); sl_send_reply("404", "User Not Found"); exit; } ######################################################################## # Request route 'base-route-local' ######################################################################## route[17] { t_on_reply("1"); if(t_check_trans()) { xlog("L_INFO", "Request leaving server - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(!t_relay()) { sl_reply_error(); } } else { xlog("L_INFO", "Dropping mis-routed request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } exit; } ######################################################################## # Request route 'base-route-generic' ######################################################################## route[18] { xlog("L_INFO", "Method not supported - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("501", "Method Not Supported Here"); exit; } ######################################################################## # Request route 'base-filter-failover' ######################################################################## route[19] { if(!t_check_status("408|500|503")) { xlog("L_INFO", "No failover routing needed for this response code - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(10); exit; } } ######################################################################## # Reply route 'base-standard-reply' ######################################################################## onreply_route[1] { xlog("L_INFO", "Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); exit; } ######################################################################## # Reply route 'base-nat-reply' ######################################################################## onreply_route[2] { xlog("L_INFO", "NAT-Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); if(nat_uac_test("1")) { fix_nated_contact(); } if(isbflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if(!search("^Content-Length:[ ]*0")) { use_media_proxy(); } } exit; } ######################################################################## # Failure route 'pstn-failover' ######################################################################## failure_route[1] { xlog("L_INFO", "Failure route for PSTN entered - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(19); if(!next_gw()) { xlog("L_ERR", "Failed to select next PSTN gateway - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(10); exit; } t_on_failure("1"); route(12); } ######################################################################## # Failure route 'base-standard-failure' ######################################################################## failure_route[2] { if(t_check_status("422|481|487")) { xlog("L_INFO", "Final reply - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(10); exit; } t_on_branch("1"); xlog("L_INFO", "Redirect from UAC intercepted - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(14); if($avp(s:cfc) != NULL) { avp_delete("$avp(s:acc_caller_user)/g"); avp_delete("$avp(s:acc_caller_domain)/g"); avp_delete("$avp(s:acc_state)/g"); avp_copy("$avp(s:acc_callee_user)", "$avp(s:acc_caller_user)"); avp_copy("$avp(s:acc_callee_domain)", "$avp(s:acc_caller_domain)"); $avp(s:acc_state) = "cfc"; avp_pushto("$ru", "$avp(s:cfc)"); setflag(29); append_branch(); t_on_branch("1"); xlog("L_INFO", "CFC detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(14); } route(19); route(10); } ######################################################################## # Branch route 'cfc-drop-local' ######################################################################## branch_route[1] { if(is_domain_local("$rd")) { xlog("L_INFO", "Dropping local branch - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); drop(); } }

1 0

[Serusers] Music on hold
by dciepiel＠orange.pl 05 Aug '07

05 Aug '07

Hi, Does anybody know how to set up music on hold with SER? I will be gratefull for any instructions how to configure this. cheers, Daniel

3 2

[OpenSER-Users] Anyone success deploy Openser with Advanced Radius Server?
by Tung Tran 04 Aug '07

04 Aug '07

1 0

Re: [Serusers] Problem with TM initialialization
by Fernando Schmitt 04 Aug '07

04 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!

1 0

[Serusers] Performance counters
by Parminder.Nayyar＠relianceada.com 04 Aug '07

04 Aug '07

Hi all Anyone here has any idea as to how we can keep the performance counters - ie the number of INVITEz recieved, the number of successfull responses to INVITE rcvd n so on... so that a performance metric cud be calculated as to how many calls got thru.. hwo many got thru voice, how many thru signalling only, n other such things??? Dint find ne module so far that cud help out in this... Has neone got ne idea?? Thanks... Parminder Nayyar

1 0

[Serusers] Problem with TM initialialization
by Fernando Schmitt 04 Aug '07

04 Aug '07

Guys, I'm 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!

1 0

Re: [OpenSER-Users] MediaProxy 2.0 roadmap
by Alexandre dos Reis Ferreira 03 Aug '07

03 Aug '07

Hi Adrian, I'd like a feature which allow get in real time the MOS of each call in the media sessions interface. Regards, Areis. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.11.2/933 - Release Date: 2/8/2007 14:22

1 0

[OpenSER-Users] openser+asterisk question
by Radu Spineanu 03 Aug '07

03 Aug '07

Hi, When deploying a multi-location openser+asterisk architecture what is the best way of storing the user accounts? I thought about separating each location with a different prefix and having a mysql DB for each one. Then tell openser to route according to the prefixes. I like this version because it means if one location fails it still works locally. And the other servers are not affected. But this brings a problem with centralizing CDR. And if I want to interconnect with other VoIP networks I would have to pick a central location, right? Thanks, Radu Spineanu

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[OpenSER-Users] Adding an attribute to SDP
by Kuddusi 03 Aug '07

03 Aug '07

Hi, How can I add a header to SDP? Actually what I want to do is, whenever an INVITE with SDP comes, search for a ptime and if ptime can not be found, add a default line to SDP body. Like if (has_body()){ if (!search_body("ptime") { # like below à # append_body("a=ptime:20"); } } Thanks, Kuddusi CIFTCIBASI Next Generation Networks Teletek Telecommunication Services Corp. Ayazma Dere Cad. Aksit Plaza No:12/1 Fulya, Besiktas , 34349, www.teletek.net Tel : +90 212 227 7030 Direkt: +90 212 310 2233 Fax : +90 212 227 8700 Email: kuddusi.ciftcibasi(a)teletek.net Skype : realkudu MSN : kuddusic(a)hotmail.com Yahoo: kuddusi.

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[OpenSER-Users] error loading presence_xml module
by Babon Frederic 03 Aug '07

03 Aug '07

Hello, I have a problem when i try to add "presence_xml" module : ERROR : load_module: could not open module (/lib/openser/modules/presence_xml.so) : /lib/openser/modules/presence_xml.so : undefined symbol : add_event Do you have any idea about this error? Here is an extract of my openser configuration file: loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" # -- presence params -- modparam("presence", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("presence_xml", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence_xml", "xcap_table", "xcaps") modparam("presence", "clean_period", 100) modparam("presence", "to_tag_pref", 'a') #modparam("presence", "lock_set_size", 8) modparam("presence", "expires_offset", 10) modparam("presence_xml", "force_active", 1) modparam("presence", "max_expires", 3600) modparam("presence", "server_address", "sip:localhost:5065") modparam("presence_xml", "pidf_manipulation", 1) Thanks for your help. Frederic.

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[Serusers] loose_route() - loose_route: no routing target is local??
by tzieleniewski 03 Aug '07

03 Aug '07

Hi, When I invoke the loose_route() function I get the following info: loose_route: no routing target is local Even though the message contains the Route header fields the functions returns false. Please point me what do I miss here?? Thanks in advance. Cheers Tomasz

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[OpenSER-Users] STUN working!!
by Marc LEURENT 03 Aug '07

03 Aug '07

Hello everybody, To remind you, I'm trying to set a SIP architecture for NATTED SIP users without using any rtp proxy (because of lack of ressource). STUN is working very well between 2 users in 2 differents NATs. But when they are behing the same NAT, they cannot reach the other.. In the INVITE and 200Ok SDP fields, they have put their public address with a port reserved with STUN: it's not working! I think that it is because it's difficult for them to reach the public IPaddress/port where they should send the RTP stream from inside the NAT. So my idea was: Why not rewriting SDP fields to put their private address when they are behind the same NAT But I don't know how to rewrite the SDP fields for both users. (I have the private address of both users in the location database) Help would be very grateful... If you have any idea! Best Regards

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[OpenSER-Users] Not possible variables or AVP's in alias_db_lookup() ?
by Iñaki Baz Castillo 03 Aug '07

03 Aug '07

I want OpenSer to dinamically look for alias in a table for each domain, so I've in my database tables with name as: dbaliases_sip1_domain_org dbaliases_sip2_domain_com dbaliases_sip3_domain_net And I do (thanks to Andreas Granig): $avp(s:dom) = $rd; avp_subst("$avp(s:dom)", "/\./_/g"); alias_db_lookup("dbaliases_" + $avp(s:dominio)); but it fails because the SQL query is literally: select username,domain from dbaliases_$avp(s:dominio) where ... I've tryed putting variables and more things into the function parameter, but it's never processed and the SQL query is always done literally. I use OpenSer 1.2.1. What can I do for solving this? Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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Re: [OpenSER-Users] NAT + STUN
by Marc LEURENT 03 Aug '07

03 Aug '07

Do you have any idea to solve this matter: In the BYE request: 0(2569) found end of header 0(2569) find_next_route: No next Route HF found 0(2569) after_loose: No next URI found So the BYE message is not forwarded by openser Please find below INVITE and BYE requests 0(2632) SIP Request: 0(2632) method: <INVITE> 0(2632) uri: <sip:103@sd-7501.dedibox.fr:5060> 0(2632) version: <SIP/2.0> 0(2632) parse_headers: flags=2 0(2632) Found param type 232, <branch> = <z9hG4bK4747925369759203710>; state=16 0(2632) end of header reached, state=5 0(2632) parse_headers: Via found, flags=2 0(2632) parse_headers: this is the first via 0(2632) After parse_msg... 0(2632) preparing to run routing scripts... 0(2632) parse_headers: flags=100 0(2632) DEBUG:parse_to:end of header reached, state=10 0(2632) DBUG:parse_to: display={}, ruri={sip:103@sd-7501.dedibox.fr:5060;user=phone} 0(2632) DEBUG: get_hdr_field: <To> [46]; uri=[sip:103@sd-7501.dedibox.fr:5060;user=phone] 0(2632) DEBUG: to body [<sip:103@sd-7501.dedibox.fr:5060;user=phone> ] 0(2632) get_hdr_field: cseq <CSeq>: <1> <INVITE> 0(2632) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(2632) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 10 0(2632) ROUTE2: STARTING NAT DETECTION 0(2632) !!!!!!!!! NAT UAC TEST 19 SUCEDEED 0(2632) parse_headers: flags=80 0(2632) ROUTE2: SETFLAG 3 0(2632) DEBUG: add_param: tag=c0a80101-b67ff5 0(2632) DEBUG:parse_to:end of header reached, state=29 0(2632) DBUG:parse_to: display={"101"}, ruri={sip:101@sd-7501.dedibox.fr:5060;user=phone} 0(2632) parse_headers: flags=200 0(2632) DEBUG: get_hdr_body : content_length=269 0(2632) found end of header 0(2632) find_first_route: No Route headers found 0(2632) loose_route: There is no Route HF 0(2632) DEBUG: has_totag: no totag 0(2632) I AM SETTING THE FLAGS FOR RADIUS 0(2632) SETTING FLAGS 1 & 2 FOR RADIUS 0(2632) CHECKING IF URI <> myself 0(2632) grep_sock_info - checking if host==us: 18==12 && [sd-7501.dedibox.fr] == [88.191.45.91] 0(2632) grep_sock_info - checking if port 5060 matches port 5060 0(2632) grep_sock_info - checking if host==us: 18==12 && [sd-7501.dedibox.fr] == [88.191.45.91] 0(2632) grep_sock_info - checking if port 5060 matches port 5060 0(2632) grep_sock_info - checking if host==us: 18==12 && [sd-7501.dedibox.fr] == [88.191.45.91] 0(2632) grep_sock_info - checking if port 5060 matches port 5060 0(2632) grep_sock_info - checking if host==us: 18==12 && [sd-7501.dedibox.fr] == [88.191.45.91] 0(2632) grep_sock_info - checking if port 5060 matches port 5060 0(2632) rewrite_uri: Rewriting Request-URI with 'sip:103@82.127.0.79:1028;user=phone' 0(2632) parse_headers: flags=ffffffffffffffff 0(2632) STARTING ROUTE 1 0(2632) subst_run: running. r=1 0(2632) subst_str: no match 0(2632) FLAG 3 OK GOTO ROUTE 3 0(2632) !!!!!!!!! ON ROUTE 3 FOR NATTED CONTACT 0(2632) DEBUG: t_newtran: T on entrance=0xffffffff 0(2632) parse_headers: flags=ffffffffffffffff 0(2632) parse_headers: flags=78 0(2632) t_lookup_request: start searching: hash=12532, isACK=0 0(2632) DEBUG: RFC3261 transaction matching failed 0(2632) DEBUG: t_lookup_request: no transaction found 0(2632) DBG: trans=0xb5c08fa8, callback type 1, id 1 entered 0(2632) trace_onreq_in: trace off... 0(2632) DBG: trans=0xb5c08fa8, callback type 1, id 0 entered 0(2632) parse_headers: flags=78 0(2632) DEBUG: noisy_timer set for accounting 0(2632) DEBUG:rr:is_direction: param ftag not found 0(2632) parse_headers: flags=ffffffffffffffff 0(2632) check_via_address(82.127.0.79, 82.127.0.79, 0) 0(2569) SIP Request: 0(2569) method: <BYE> 0(2569) uri: <sip:101@82.127.0.79:1312> 0(2569) version: <SIP/2.0> 0(2569) parse_headers: flags=2 0(2569) Found param type 232, <branch> = <z9hG4bK2074253192092946047>; state=16 0(2569) end of header reached, state=5 0(2569) parse_headers: Via found, flags=2 0(2569) parse_headers: this is the first via 0(2569) After parse_msg... 0(2569) preparing to run routing scripts... 0(2569) parse_headers: flags=100 0(2569) DEBUG: add_param: tag=c0a80101-b31387 0(2569) DEBUG:parse_to:end of header reached, state=29 0(2569) DBUG:parse_to: display={}, ruri={sip:101@sd-7501.dedibox.fr:5060;user=phone} 0(2569) DEBUG: get_hdr_field: <To> [66]; uri=[sip:101@sd-7501.dedibox.fr:5060;user=phone] 0(2569) DEBUG: to body [<sip:101@sd-7501.dedibox.fr:5060;user=phone>] 0(2569) get_hdr_field: cseq <CSeq>: <1> <BYE> 0(2569) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(2569) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 10 0(2569) ROUTE2: STARTING NAT DETECTION 0(2569) !!!!!!!!! NAT UAC TEST 19 SUCEDEED 0(2569) parse_headers: flags=80 0(2569) DEBUG: get_hdr_body : content_length=0 0(2569) found end of header 0(2569) ROUTE2: SETFLAG 3 0(2569) DEBUG: add_param: tag=c0a80101-2376fc2 0(2569) DEBUG:parse_to:end of header reached, state=29 0(2569) DBUG:parse_to: display={}, ruri={sip:103@sd-7501.dedibox.fr:5060;user=phone} 0(2569) parse_headers: flags=200 0(2569) is_preloaded: No 0(2569) grep_sock_info - checking if host==us: 11==12 && [82.127.0.79] == [88.191.45.91] 0(2569) grep_sock_info - checking if port 5060 matches port 1312 0(2569) grep_sock_info - checking if host==us: 11==12 && [82.127.0.79] == [88.191.45.91] 0(2569) grep_sock_info - checking if port 5060 matches port 1312 0(2569) DEBUG:check_self: host != me 0(2569) grep_sock_info - checking if host==us: 12==12 && [88.191.45.91] == [88.191.45.91] 0(2569) grep_sock_info - checking if port 5060 matches port 5060 0(2569) after_loose: Topmost route URI: 'sip:88.191.45.91;lr=on;ftag=c0a80101-b31387' is me 0(2569) parse_headers: flags=200 0(2569) found end of header 0(2569) find_next_route: No next Route HF found 0(2569) after_loose: No next URI found 0(2569) DBG:rr:run_rr_callbacks: callback id 0 entered with <lr=on;ftag=c0a80101-b31387> Dan-Cristian Bogos a écrit : > That's because for INVITE your will take routing decisions but BYE u > will just proxy out. > > DanB > > On 8/2/07, Marc LEURENT <lftsy(a)free.fr> wrote: >> I've compared the INVITE and BYE method.... >> And the uri in the INVITE method is >> 0(2632) SIP Request: >> 0(2632) method: <INVITE> >> 0(2632) uri: <sip:103@sd-7501.dedibox.fr:5060> >> 0(2632) version: <SIP/2.0> >> >> whereas in the BYE method >> >> 0(2569) SIP Request: >> 0(2569) method: <BYE> >> 0(2569) uri: <sip:101@82.127.0.79:1312> >> 0(2569) version: <SIP/2.0> >> >> >> so the DEBUG:check_self: host != me >> I'm going to try without accoounting, but it should'nt change anything... >> >> Best Regards >> >> >> >> >> >>

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Re: [OpenSER-Users] how does OpenSER run on IPV6?
by Daniel-Constantin Mierla 03 Aug '07

03 Aug '07

Hello, On 08/03/07 08:41, null_neo(a)sina.com wrote: > Hi > > I am glad to recieve a mail to help me! > I have worked out today.Just force OpenSER only listening on IPV6 address. yep, maybe should be stated clear that you should force listen=ipv6_address to have openser managing ipv6 traffic. Daniel > And I use Linphone as UA ,they worked well. > HeHe,I'm very happy because I finished this job(without reward) and I > can have my holidays some days later. > Thank you. > > > > ----- Original Message ----- > From:Daniel-Constantin Mierla > To:null_neo@sina.com > Subject:Re: [OpenSER-Users] how does OpenSER run on IPV6? > Date:07-08-02 18:11:37 > > Hello, > > the "result was bad" does not give us hints to help you. > Please paste > the error messages you get. > > Cheers, > Daniel > > > On 07/31/07 04:29, null_neo(a)sina.com wrote: > > > > Dear sir: > > > > First,I am a Chinese,and my English is poor.^_^ > > > > We are testing kinds of applications on IPV6 including VOIP. And we > > use Linphone as our UA. If I don't use domain name,how can I add a > > account? On IPV4,I add account " # openserctl add test test > > test@ipv4_address" .And in the file "openser.cfg" ,I > directly write my > > SIP server ipv4_address . Can I directly replace > "ipv4_address" by > > "ipv6_address"? I has tried,and the resault is bad. > > > > I am very sorry that I am amateurish for VOIP ,because Our work is > > only to test. > > > > Thank you! > > > > Li Zhonglei > > > > > > > > ------------------------------------------------------------------- > > �� ҵ��ܵ��( > > > http://d1.sina.com.cn/sina/limeng3/mail_zhuiyu/2007/mail_zhuiyu_20070723.ht… > > > ) > > > > =================================================================== > > ע��2G��䣨 http://mail.sina.com.cn/chooseMode.html �� > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > ------------------------------------------------------------------- > �� ҵ��ܵ��( > http://d1.sina.com.cn/sina/limeng3/mail_zhuiyu/2007/mail_zhuiyu_20070723.ht… > ) > > =================================================================== > ע��2G��䣨 http://mail.sina.com.cn/chooseMode.html ��

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[OpenSER-Users] SCTP support
by Daniel-Constantin Mierla 03 Aug '07

03 Aug '07

Hello, courtesy of Glenn Marmon, Connecticut College Class of 2007, who submitted the patch, OpenSER includes now support for SCTP (Stream Control Transmission Protocol). It completes the list of the possible IP transport layers that could be used with SIP in OpenSER. You can find details about SCTP at: http://www.sctp.org/ Basically, it is a reliable transport protocol, like TCP, but provides a number of functions that are critical for telephony signaling transport (read more about the benefits: http://www.ietf.org/rfc/rfc3286.txt). If you have the chance to own a SCTP-aware SIP phone and can test, please provide us feedback on the mailing lists -- also, let us know the name and type of any SIP phone with SCTP support that you know. Hope SCTP will gain more and more space, as it seems to be the right protocol for VoIP/IM&P signaling, where UDP cannot cope very well with huge messages and TCP adds too much overhead. Cheers, Daniel

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[OpenSER-Users] Logs to /var/log/openser
by Iñaki Baz Castillo 03 Aug '07

03 Aug '07

Hi, I've configured: log_facility=LOG_LOCAL0 and added in "syslog.conf": # don't log messages with LOG_LOCAL0 in /var/log/syslog anymore *.*;auth,authpriv.none,local0.none -/var/log/syslog # # log messages with LOG_LOCAL0 in /var/log/ser.log local0.* -/var/log/ser.log as it appears in http://docs.huihoo.com/openser/tutorials/ser-syslog/ar01s06.html OpenSer then logs in /var/log/openser, true, but it still logs in /var/log/syslog. ¿?¿ Obviously I don't want OpenSer logs appearing in /var/log/syslog. I use Debian Etch, any idea of what can be wrong? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] avp_subst and regex
by Zahid Mehmood 02 Aug '07

02 Aug '07

Hi, I am trying to rewrite a 5 digit number to a 11 digit number by adding a prefix "121285". To achieve that I use the following: avp_subst("$avp(s:fromuri)","/^(sip:)([134][0-9]{4}@.*)/\1+121285\2/"); in openser.log new fromuri = sip:+12128512345@host.columbia.edu It works fine because of the "+" after \1 This avp_subst does not work properly if the "+" is taken out. How can I use "\1" followed by another digit? avp_subst("$avp(s:fromuri)","/^(sip:)([134][0-9]{4}@.*)/\1121285\2/"); in openser.log: new fromuri = J128512345(a)host.columbia.edu What am i doing wrong? Thanks in advance for your help. -- Zahid

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[OpenSER-Users] OpenSER looping routes back to itself.
by Taylor Carpenter 02 Aug '07

02 Aug '07

Running OpenSER v1.1.1 with internal ip for sip client connections and ext ip to connect to sip providers. I am getting loops in my calls sometimes... They stop after reaching the max hops trying to send the BYE message to it self. I am wondering if it is the record- route that I see that includes the internal IP. The 483 error is U 2007/07/30 00:49:56.133157 192.168.18.22:5060 -> 192.168.18.22:5060 SIP/2.0 483 Too Many Hops. From: <sip:15126461516@192.168.18.1:5060>;tag=nu-481d-6c62. To: <sip:18065436468@192.168.18.22:5060>;tag=6738. Call-ID: 421ac0c6-1dd2-11b2-8093-df998124da20(a)192.168.18.91. CSeq: 11269143 BYE. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.7fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.6fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.5fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.4fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.3fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.2fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.1fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.0fe832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.fee832a5.0. Via: SIP/2.0/UDP 192.168.18.22;rport=5060;branch=z9hG4bKe6ff.eee832a5.0. Via: SIP/2.0/UDP 192.168.18.91:5060;rport=5060. Server: OpenSer (1.1.1-notls (i386/linux)). Content-Length: 0. Any ideas?

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[OpenSER-Users] benchmark module
by Daniel-Constantin Mierla 02 Aug '07

02 Aug '07

Hello, starting from the original patch submitted by Bastian Friedrich and incorporating the code used in the core for past benchmarking, the devel version has a new module "benchmark". It can be used to log the duration of a configuration part execution, allowing in this way to detect slow elements of routing script. I am sure is a important tool, at least in preparing production systems, so development will continue to make more flexible the printed statistics. The readme file will give you more details: http://www.openser.org/docs/modules/1.3.x/benchmark.html The module exports an internal API, so developers can do benchmarking inside the C code. Feedback is appreciated. Cheers, Daniel

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Re: [OpenSER-Users] Replacing "." of a domain with "_" and store the result in a variable
by Iñaki Baz Castillo 02 Aug '07

02 Aug '07

El Thursday 02 August 2007 17:31:23 escribió: > Hi, > > $avp(s:dom) = $rd; > avp_subst("$avp(s:dom)", "/\./_/g"); Great, didn't realize of that :) Thanks a lot. > > Andreas > > Iñaki Baz Castillo wrote: > > Hi, I just need to take the requested domain $rd and store in a variable > > that domain but with . replaced by _: > > > > sip.domain.org -> sip_domain_org > > > > This is because I want tables in MySQL containing the domain name but > > MySQL don't allow . in table name. > > > > So now, what should I do for that? I really thought "textops" could help > > me but I don't see a function to replace text and store it in a variable. > > > > I see "replace()" and "subst()" but they seem to exist for replacing text > > in the SIP message "on the fly", that is not that I'm looking for. > > > > Any suggestion? Thanks a lot. -- ilimit... *Iñaki Baz Castillo* ibc(a)in.ilimit.es ÀREA SISTEMES 0034 937 333 375 VOLTA 1, PIS 5 08224 TERRASSA.BCN Aquest enviament és confidencial i està destinat únicament a la persona a qui s'ha enviat. Pot contenir informació privada sotmesa al secret professional, la distribució de la qual està prohibida per la legislació vigent.

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[OpenSER-Users] Replacing "." of a domain with "_" and store the result in a variable
by Iñaki Baz Castillo 02 Aug '07

02 Aug '07

Hi, I just need to take the requested domain $rd and store in a variable that domain but with . replaced by _: sip.domain.org -> sip_domain_org This is because I want tables in MySQL containing the domain name but MySQL don't allow . in table name. So now, what should I do for that? I really thought "textops" could help me but I don't see a function to replace text and store it in a variable. I see "replace()" and "subst()" but they seem to exist for replacing text in the SIP message "on the fly", that is not that I'm looking for. Any suggestion? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] OPENSERAMIN:SocketError in loginController#login
by zze-WU Yin RD-MAPS-LAN 02 Aug '07

02 Aug '07

Hello everybody, I haved installed a openseradmin. And I can have a page for login in the http://localhost:3000. But when i loggined with: login: openser password:openserrw I have got a page of error : SocketError in loginController # login Do anybody know why? Thank you very much! Yin

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[OpenSER-Users] NAT + STUN
by Marc LEURENT 02 Aug '07

02 Aug '07

Hello evrybody, I need a piece of advice if you have time.... Thanks I have some trouble with my configuration file, I'm trying to set a SIP network working through NAT using STUN. REGISTRATION is working, INVITE are working, I can set a communication with 2 working phones but sip pings an BYE messages are not forwared... Any idea?? Thanks # U 88.191.45.91:5060 -> 82.127.0.79:1312 OPTIONS sip:82.127.0.79:1312 SIP/2.0. Via: SIP/2.0/UDP 88.191.45.91:5060;branch=0. From: sip:pinger@sip.leurent.eu;tag=804466c1. To: sip:82.127.0.79:1312. Call-ID: 23ce8297-c09c8eb7-a(a)88.191.45.91. CSeq: 1 OPTIONS. Content-Length: 0. . # U 82.127.0.79:1312 -> 88.191.45.91:5060 SIP/2.0 481 CallLeg/Transaction Does Not Exist. Via: SIP/2.0/UDP 88.191.45.91:5060;branch=0. From: <sip:pinger@sip.leurent.eu>;tag=804466c1. To: <sip:82.127.0.79:1312>;tag=c0a80101-4bf1d8. Call-ID: 23ce8297-c09c8eb7-a(a)88.191.45.91. CSeq: 1 OPTIONS. Content-Length: 0. . # U 82.127.0.79:1312 -> 88.191.45.91:5060 BYE sip:103@82.127.0.79:1027 SIP/2.0. Via: SIP/2.0/UDP 82.127.0.79:1313;branch=z9hG4bK8030359792092547043. From: "101"<sip:101@sip.leurent.eu:5060;user=phone>;tag=c0a80101-4c5eed. To: <sip:103@sip.leurent.eu:5060;user=phone>;tag=c0a80101-1d0bb0d. Call-ID: 66464a0-c0a80101-0-1f(a)192.168.95.4. CSeq: 2 BYE. Max-Forwards: 70. Route: <sip:88.191.45.91:5060;lr=on;ftag=c0a80101-4c5eed>. User-Agent: THOMSON ST2030 hw0 fw1.50 00-0E-50-4E-AF-AE. Content-Length: 0. . # # $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=7 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) children=4 #alias=sip.leurent.eu port=5060 #server_signature=yes #tos=IPTOS_LOWDELAY avp_aliases="day=i:101;time=i:102;can_uri=i:800;s_ip=i:801;billing_party=i:802;from_header=i:803;sip_proxy_ip=i:804" #;pstnuser=i:805;pstnpassword=i:806:pstnrealm=i:807" # ------------------ module loading ---------------------------------- #set module path mpath="/usr/lib/openser/modules/" # Uncomment this if you want to use SQL database loadmodule "mysql.so" loadmodule "sl.so" # Stateless Module loadmodule "tm.so" # Transaction Module loadmodule "rr.so" # Record-Route and Route Module loadmodule "maxfwd.so" # Max-Forward processor Module loadmodule "usrloc.so" # User Location Implementation Module loadmodule "registrar.so" # SIP Registrat Implementation Module (need usrloc) loadmodule "textops.so" # Text Operation Module loadmodule "mi_fifo.so" # FIFO transport layer implementation for Management Interface loadmodule "acc.so" # Accounting Module loadmodule "avpops.so" # AVP Operation Module (user preference) loadmodule "uri.so" # Generic URI operation Module loadmodule "auth.so" # Authentification Module #loadmodule "auth_db.so" # Database-backend Authentication mMdule loadmodule "auth_radius.so" # RADIUS-backend Authentication Module loadmodule "group_radius.so" # User-groups Module with RADIUS-backend #loadmodule "avp_radius.so" # RADIUS-backend for AVP loading Module #loadmodule "presence.so" # Presence server Module #loadmodule "pua.so" # Common API for presence user agent client loadmodule "options.so" # OPTIONS server replier Module loadmodule "xlog.so" # Advanced Logger Module loadmodule "nathelper.so" # NAT Traversal Helper Module #loadmodule "dispatcher.so" # Dispatcher (load-balancer) Module loadmodule "uac.so" # User Agent Client loadmodule "siptrace.so" # SipTrace module (storage of SIP requests) #loadmodule "exec.so" # Allows to start an external command from a OpenSER script # ----------------- setting module-specific parameters --------------- # -- maxfwd params -- modparam("maxfwd", "max_limit", 10) # Default is 256 | 10 in the functions # -- sl params -- #modparam("sl", "enable_stats", 1) # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- # Uncomment this if you want to use SQL database modparam("usrloc", "db_mode", 1) # Write instantaneously in the DB modparam("usrloc", "db_url", "mysql://openser:test@127.0.0.1/openser") modparam("usrloc", "timer_interval", 10) modparam("usrloc", "nat_bflag" , 3) # -- rr params -- modparam("rr", "enable_full_lr", 1) # add value to ;lr param to make some broken UAs happy # -- siptrace params -- modparam("siptrace", "db_url", "mysql://openser:test@127.0.0.1/openser") modparam("siptrace", "table", "sip_trace") # Default value "sip_trace" modparam("siptrace", "trace_on", 1) # -- registrar params -- modparam("registrar", "default_expires", 1800) modparam("registrar", "received_avp", "$avp(i:42)") # -- nathelper params -- modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "sipping_bflag", 5) modparam("nathelper", "natping_interval", 10) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "received_avp", "$avp(i:42)") # Same Value as the registrar module modparam("nathelper", "sipping_from", "sip:pinger@sip.leurent.eu") # -- auth params -- #modparam("auth", "secret", "johndoessecretphrase") # Default is random => don't set it #modparam("auth", "nonce_expire", 300) # Time before nounce expiration modparam("auth_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") # -- group_radius params -- modparam("group_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("group_radius", "use_domain", 1) # username@domain will be used for lookup # -- avp_radius parameter -- #modparam("avp_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") # -- acc params (with radius )-- modparam("acc", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) modparam("acc", "early_media", 1) modparam("acc", "report_cancels", 1) #modparam("acc", "report_ack", 0) modparam("acc", "detect_direction", 1) #modparam("acc", "log_flag", 1) # number of the flag which will be used to mark messages for accounting #modparam("acc", "log_level", 1) # Set the reporting log level #modparam("acc", "log_missed_flag", 2) # #modparam("acc", "failed_transaction_flag", 2) modparam("acc", "service_type", 15) # Radius service type used for accounting : 15 = (SIP) #modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp") # ATTENTION: DO NOT PUT ; at the end of the radius_extra attribute modparam("acc", "radius_extra", "Sip-Src-IP=$si; Sip-Src-Port=$sp; Canonical-URI=$avp(can_uri); Billing-Party=$avp(billing_party); SIP-Proxy-IP=$avp(sip_proxy_ip); User-Agent=$ua ") #Billing-Party=$avp(billing_party) #From-Header=$hdr(from); #User-Name=$fU; #From-Header=$avp(from_header); #Digest-Realm=$fd #Sip-From-Tag=$avp(from_header); #SIP-Method=$rm; # ------------------------- 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; }; # NAT detection route(2); # 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"); if(is_method("BYE")) { # log it all the time acc_rad_request("200 ok"); acc_log_request("200 ok"); } route(1); }; # Set the acc flags if(is_method("INVITE") && !has_totag()) { xlog("L_INFO", "I AM SETTING THE FLAGS FOR RADIUS \r\n"); $avp(can_uri) = $ru; # SIP Request's URI $avp(billing_party) = $fu; # From URI $avp(from_header) = $fU; # From URI username $avp(sip_proxy_ip) = $Ri; # Received IP address setflag(1); # radius_flag setflag(2); # radius_missed_flag }; # Functions when calling other domains if (!uri==myself) { # check if user is allowed to do voip calls to other domains # if(is_method("INVITE|MESSAGE")) { # if (radius_is_user_in("From", "voip")) { # sl_send_reply("403", "Forbidden VoIP"); # exit; # }; # }; # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { sip_trace(); xlog("L_INFO", "$fU IS TRYING TO REGISTER \r\n"); if (!radius_www_authorize("sip.leurent.eu")) { www_challenge("sip.leurent.eu", "0"); # qop set to 1 xlog("L_INFO", "WWW_CHALLENGE of $si FAILED \r\n"); exit; }; #if (isflagset(5)) { if (isbflagset(3)) { #setflag(6); # if you want OPTIONS natpings uncomment next # setflag(7); # Deprecated setbflag(5); # Set Flag for SIP PINGING }; save("location"); xlog("L_INFO", "SAVE LOCATION OF $si \r\n"); exit; }; if (!lookup("location")) { # log to acc as missed call acc_rad_request("404 Not Found"); acc_log_request("404 Not Found"); xlog("L_DBG", "ACC RADIUS: 404 NOT FOUND FOR $si \r\n"); sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } ## Generic Forward route[1] { if (subst_uri('/(sip:.*);nat=yes/\1/')){ #setflag(6); # Deprecated, for version 1.1 setbflag(3); }; #if (isflagset(5)||isflagset(6)) { if (isbflagset(3)) { route(3); } if (!t_relay()) { sl_reply_error(); }; exit; } # NAT Detection route[2]{ force_rport(); if (nat_uac_test("19")) { xlog("!!!!!!!!! NAT UAC TEST 19 SUCEDEED \r\n"); if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; #setflag(5); Deprecated setbflag(3); }; } ## Route for natted contact route[3] { if (is_method("BYE|CANCEL")) { # Ajout Maison acc_rad_request("200 ok"); acc_log_request("200 ok"); #unforce_rtp_proxy(); } else if (is_method("INVITE")){ #force_rtp_proxy(); t_on_failure("1"); }; #if (isflagset(5)) if (isbflagset(3)){ search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); } t_on_reply("1"); } ## Failure Route 1 failure_route[1] { xlog("!!!!!!!!! ON FAILURE ROUTE \r\n"); #if (isflagset(6) || isflagset(5)) { if (isbflagset(3)) { #unforce_rtp_proxy(); } } ## Reply route onreply_route[1] { xlog("!!!!!!!!! ON REPLY ROUTE \r\n"); #if ((isflagset(5) || isflagset(6)) && status=~"(183)|(2[0-9][0-9])") { if (isbflagset(3) && status=~"(183)|(2[0-9][0-9])") { #force_rtp_proxy(); } search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); #if (isflagset(6)) { if (isbflagset(3)) { xlog("!!!!!!!!! ON REPLY ROUTE / FIX NATED CONTACT \r\n"); fix_nated_contact(); } exit; }

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[OpenSER-Users] openser to solve SIP+RTP+NAT problems in typical SOHO setup
by Jens Thiele 02 Aug '07

02 Aug '07

Hi, I am evaluating software to solve SIP+RTP+NAT problems in typical SOHO setups: Scenarios: Typical scenarios are (A)DSL "dial-up / dynamic IP" connections with one public IP and a router doing NAT for some LANs. 1) WAN multiple LANs internet <-> NAT router <-> SIP phones sip registrar SIP proxy <-> SIP phones with NAT logic : 2) WAN multiple LANs internet <-> NAT router <-> SIP proxy with NAT logic, SIP phones sip registrar <-> SIP phones : Notes: - the SIP phones use the SIP proxy with NAT logic as outbound proxy - the proxy is notified if the public IP changes The closest match so far has been siproxd [1]. But I would prefer to use openser+nathelper+rtpproxy. Now it seems nathelper is designed for a different scenario (SIP provider): WAN/internet sip registrar <-> ser+nathelper <-> NAT router <-> multiple LANs with SIP phones and uses the packet source address in fix_nated_contact. Now I stumbled across milkfish [2] and its patches [3] to openser (nathelper). It seems the patch allows to explicitly set the address used by fix_nated_contact to rewrite the contact header (instead of using the packet source address). Searching the mailing list archives I did not find a better solution. Closest match I found was the thread: "[Devel] Extended nathelper" started by Walter Schober http://www.mail-archive.com/devel@openser.org/msg01930.html Ending in the suggestion to use add_rvc_param(). But add_rcv_param seems to use the packet source address, too. But also from this thread it seems one might use the mangle module to rewrite the contact header field, but then I am not sure about the state of this module: "This is a module to help with SDP mangling. Still in testing." I am a bit surprised that there is no straight forward solution to such a common setup. Am I missing something really obvious? Thanks Greetings Jens Thiele [1] siproxd <http://siproxd.sourceforge.net/> [2] milkfish <http://wiki.milkfish.org> [3] milkfish openser patches <http://packages.milkfish.org/boozy/Milkfish_Sources_for_OpenWrt-SDK/OpenWrt…>

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[OpenSER-Users] openser+radius
by ram 02 Aug '07

02 Aug '07

Hi iam installing openser SVN and Freeradius 1.16 followed the link as mentioned in document when i run radiusd -X iam getting below problem 1. rlm_sql (sql): Connected new DB handle, #4 2. Module: Instantiated sql (sql) 3. Module: Loaded Acct-Unique-Session-Id 4. acct_unique: key = "User-Name, Acct-Session-Id, NAS-IP-Address, Client-IP-Address, NAS-Port" 5. rlm_acct_unique: Cannot find attribute 'NAS-IP-Address' in dictionary 6. radiusd.conf[1159]: acct_unique: Module instantiation failed. 7. radiusd.conf[1753] Unknown module "acct_unique". 8. radiusd.conf[1747] Failed to parse preacct section. any suggestiong ram

2 1

[OpenSER-Users] PRACK handling
by Zahid Mehmood 02 Aug '07

02 Aug '07

Hello everyone, What is the preferred way of handling PRACKS? simply relay them if they are in loose_route? should they be authenticated? What, if any, are the potential risks of not authenticating PRACKS? Thanks in advance. -- Zahid

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[OpenSER-Users] disable media/nathelper
by Aymeric Moizard 02 Aug '07

02 Aug '07

Hi, I'm facing an issue with openser. I checked the latest SVN code and it seems my issue is not solved even in the svn version. My UA is sending an SDP with 2 m lines: one for audio and one for video. In a re-INVITE, I want to disable video and I'm sending m=video 0 RTP/AVP.. My openser is currently modifying the port number for the video line even if it's 0 which means it's disabled... This behavior would of course happen even for audio lines that ones wants to disable. I was not able to find a clean fix in nathelper.c and hope to get some help for proposing one... I guess it would be something close to: if (oldport==0) skip altering ip/port for media and continue; but I don't want to insert any mem leak or lose a session... tks, Aymeric MOIZARD / ANTISIP amsip - http://www.antisip.com osip2 - http://www.osip.org eXosip2 - http://savannah.nongnu.org/projects/exosip/

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[OpenSER-Users] how does OpenSER run on IPV6?
by null_neo＠sina.com 02 Aug '07

02 Aug '07

Dear sir: First,I am a Chinese,and my English is poor.^_^ We are testing kinds of applications on IPV6 including VOIP. And we use Linphone as our UA. If  I don't use domain name,how can I add a account? On IPV4,I add account " # openserctl add test test test@ipv4_address" .And in the file "openser.cfg" ,I directly write my SIP server ipv4_address . Can I directly replace "ipv4_address" by "ipv6_address"?  I has tried,and the resault is bad. I am very sorry that I am amateurish for VOIP ,because Our work is only to test. Thank you!                                                                                                 Li  Zhonglei ------------------------------------------------------------------- 吉贝克 ——商业智能的领军者( http://d1.sina.com.cn/sina/limeng3/mail_zhuiyu/2007/mail_zhuiyu_20070723.ht… ) =================================================================== 注册新浪2G免费邮箱（ http://mail.sina.com.cn/chooseMode.html ）

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[OpenSER-Users] about openSER and SIPp
by bozhang＠nec-labs.com 02 Aug '07

02 Aug '07

Hi there, I'm doing sipp performance measurements with openser 1.2. I run OpenSER on machine A, then run one SIPp on machine B, one SIPp on machine C. For the SIPp on machine C, I start it using UAS mode, ie., ./sipp -sn uas For the SIPp on machine B, if I start is using the following command: ./sipp -sn uac C -rsa A -m 1 -r 1 Everything is correct. The OpenSER can relay messages correctly. But if I start SIPp on machine B using the following command: ./sipp -sn uac -rsa A -m 1 -r 1 -d 5000 Then the SIPp client on machine B will crash because of receiving unexpected "ringing 180" message. Note: the only difference is I add a pause 5 seconds using "-d 5000" option. What I found is: the OpenSER server will keep sending "INVITE" to C during the '5 second pause', so C will keep reponding with "180 Ringing" to B. Does anybody know why OpenSER keeps sending "INVITE" to C while B only issues one "INVITE" request? BTW, the configuration file for OpenSER is very simple: # # $Id: openser.cfg 1676 2007-02-21 13:16:34Z 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=32 log_facility=LOG_LOCAL0 disable_tcp=yes disable_dns_blacklist=yes disable_dns_failover=yes # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes # listen=udp:192.168.2.102:5060 # ------------------ module loading ---------------------------------- loadmodule "modules/tm/tm.so" modparam("tm", "wt_timer", 2) # ------------------------- request routing logic ------------------- # main routing logic route{ t_relay(); } Thanks a lot! -Bo Zhang

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Re: [OpenSER-Users] Dispatcher
by Daniel-Constantin Mierla 02 Aug '07

02 Aug '07

OK, so you didn't get in the condition of the failure route. Daniel On 08/02/07 12:58, Thomas Deillon wrote: > Hello, > > It's ok, I finally find the solution with raviprakash sunkara. > It was the parameter of tm. fw_timer. > > Cheers, > > Thomas > > On 02/08/07, *Daniel-Constantin Mierla * <daniel(a)voice-system.ro > <mailto:daniel@voice-system.ro>> wrote: > > Hello, > > can you run in debug=7 and log_stderror=yes and watch the debug > messages? If you have issues understanding what is happening, then > just > forward it to me along with dispatcher.list file. > > Daniel > > On 08/01/07 15:56, Thomas Deillon wrote: > > Hi, > > > > I'm using OpenSER as PROXY and REGISTRAR server, a Cisco as Gateway > > and 2 Asterisks (using dispatcher) as Voicemail and conference > Servers. > > > > It works perfectly on load-balancing, but not in failover because I > > never go on the failure route. > > Do you have a idea (a part of my config file below) ? > > rq: Just for information, the 12 is the milliwatt() function on > Asterisks. > > > > my openser.cfg > > > > modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") > > modparam("usrloc", "db_url", "mysql://openser:openser@localhost > /openser") > > modparam("usrloc", "db_mode", 1) > > > > > modparam("dispatcher","list_file","/usr/local/etc/openser/dispatcher.list") > > modparam("dispatcher","flags",2) > > modparam("dispatcher", "dst_avp", "$avp(i:271)") > > modparam("dispatcher", "grp_avp", "$avp(i:272)") > > modparam("dispatcher", "cnt_avp", "$avp(i:273)") > > modparam("dispatcher", "force_dst", 1) > > > > route { > > [...] > > > > if (method=="INVITE") > > { > > if (uri=~"sip:12@.*") > > { > > xlog( "L_ERR", "A: $rm $fu:$op -> $du AVP: $avp(i:271), > > $avp(i:272), $avp(i:273)\n" ); > > ds_select_dst("1", "4"); > > #t_relay(); > > xlog( "L_ERR", "B: $rm $fu:$op -> $du AVP: $avp(i:271), > > $avp(i:272), $avp(i:273)\n" ); > > t_on_failure("1"); > > } > > } > > [...] > > } > > > > failure_route[1] > > { > > xlog( "L_ERR", "C: $rm $fu:$op -> $du\n" ); > > if(t_check_status("408")) > > { > > ds_mark_dst(); > > ds_next_dst(); > > } > > else { t_reply("501", "Not Implemented"); } > > > > xlog( "L_ERR", "D: $rm $fu:$op -> $du\n" ); > > } > > > > Cheers, > > > > Thomas > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org <mailto:Users@openser.org> > > http://openser.org/cgi-bin/mailman/listinfo/users > > > >

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[OpenSER-Users] Dispatcher
by Thomas Deillon 02 Aug '07

02 Aug '07

Hi, I'm using OpenSER as PROXY and REGISTRAR server, a Cisco as Gateway and 2 Asterisks (using dispatcher) as Voicemail and conference Servers. It works perfectly on load-balancing, but not in failover because I never go on the failure route. Do you have a idea (a part of my config file below) ? rq: Just for information, the 12 is the milliwatt() function on Asterisks. my openser.cfg modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("usrloc", "db_url", "mysql://openser:openser@localhost/openser") modparam("usrloc", "db_mode", 1) modparam("dispatcher","list_file","/usr/local/etc/openser/dispatcher.list") modparam("dispatcher","flags",2) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "grp_avp", "$avp(i:272)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("dispatcher", "force_dst", 1) route { [...] if (method=="INVITE") { if (uri=~"sip:12@.*") { xlog( "L_ERR", "A: $rm $fu:$op -> $du AVP: $avp(i:271), $avp(i:272), $avp(i:273)\n" ); ds_select_dst("1", "4"); #t_relay(); xlog( "L_ERR", "B: $rm $fu:$op -> $du AVP: $avp(i:271), $avp(i:272), $avp(i:273)\n" ); t_on_failure("1"); } } [...] } failure_route[1] { xlog( "L_ERR", "C: $rm $fu:$op -> $du\n" ); if(t_check_status("408")) { ds_mark_dst(); ds_next_dst(); } else { t_reply("501", "Not Implemented"); } xlog( "L_ERR", "D: $rm $fu:$op -> $du\n" ); } Cheers, Thomas

2 1

[OpenSER-Users] Header Mangling
by Peter Gradwell 02 Aug '07

02 Aug '07

hi ya, When we send phone call out to our PSTN gateways and carriers, we often have to modify the To and From headers to put them in the right syntax and to copy the caller id up from the remote-party-id header. We think the uac module will allow us to do this on the From header, but we also need to add/replace digits in the To header, e.g. changing it into INTL format, from 01225 to 441225... If we do this using TextOps then the problem is that it only changes the string sent to the Telco switch, and it is not therefore changed on the reply, and so the end user agent does not then handle the replies properly. Does anyone know how we can mangle the To header? Is there a similar module to uac for To headers? cheers peter

2 1

[OpenSER-Users] Attribue User-name was not found in request, unique ID MAY be inconsistent
by zze-WU Yin RD-MAPS-LAN 02 Aug '07

02 Aug '07

Hi, everybody Another probleme with Openser and Freeradius. When I tried to use the acc module, i recieved: rlm_acct_unique: WARNING Attribue User-name was not found in request, unique ID MAY be inconsistent And in the Accounting-Request packet, i can not find User-name. Do anybody know why? Thank you very much!!

2 1

[OpenSER-Users] Pb with log_fmt
by zze-WU Yin RD-MAPS-LAN 02 Aug '07

02 Aug '07

Hi, thank you very much. I know what the problem is now. Because I use a openser-1.2.1-notls. So the "log_fmt" is obsolete. In the http://www.openser.org/docs/modules/1.2.x/acc.htm it wrote that: The accouting module will log by default a fixed set of attributes for the transaction - if you customize you accouting by adding more information to be logged, please see the next chapter about extra accouting - Section 1.2. The fixed minimal accouting information is: Request Method name >From header TAG parameter To header TAG parameter Call-Id 3-digit Status code from final reply Reason phrase from final reply Time stamp when transaction was completed If a value is not present in request, the empty string is accounted instead. I want have a User-Name in the accounting information. So I saw in the Section 1.2 Selection of extra information is done via xxx_extra parameters by specifying the names of additional information you want to log. This information is defined via pseudo-variables and may include headers or AVPs values or other message or system values. The syntax of the parameter is: xxx_extra = extra_definition (';'extra_definition)* extra_definition = log_name '=' pseudo_variable And the pseudo_variable I need is '$au' for User-Name. But I don't understand how to use this paramater in the openser.cfg. Do you have some ideas??

2 1

[OpenSER-Users] Fw: Adding an attribute to SDP
by Kuddusi 01 Aug '07

01 Aug '07

Hi, How can I add a header to SDP? Actually what I want to do is, whenever an INVITE with SDP comes, search for a ptime and if ptime can not be found, add a default line to SDP body. Like if (has_body()){ if (!search_body("ptime") { # like below à # append_body("a=ptime:20"); } } Thanks, Kuddusi CIFTCIBASI Next Generation Networks Teletek Telecommunication Services Corp. Ayazma Dere Cad. Aksit Plaza No:12/1 Fulya, Besiktas , 34349, www.teletek.net Tel : +90 212 227 7030 Direkt: +90 212 310 2233 Fax : +90 212 227 8700 Email: kuddusi.ciftcibasi(a)teletek.net Skype : realkudu MSN : kuddusic(a)hotmail.com Yahoo: kuddusi.

1 0

[OpenSER-Users] Adding a header to SDP
by Kuddusi CIFTCIBASI 01 Aug '07

01 Aug '07

Hi, How can I add a header to SDP? Actually what I want to do is, whenever an INVITE with SDP comes, search for a ptime and if ptime can not be found, add a default header to SDP body. Like if (has_body()){ if (search_body("ptime") { # like below --> # append_body("a=ptime:20"); } } Thanks, Kuddusi CIFTCIBASI Next Generation Networks Teletek Telecommunication Services Corp. Ayazma Dere Cad. Akşit Plaza No:12/1 Fulya, Beşiktaş , 34349, www.teletek.net Tel : +90 212 227 7030 Direkt: +90 212 310 2233 Fax : +90 212 227 8700 Email: kuddusi.ciftcibasi(a)teletek.net Skype : realkudu MSN : kuddusic(a)hotmail.com <mailto:kuddusic@hotmail.com> Yahoo: kuddusi. Yasal Uyari: Bu e-posta'nin içerdigi bilgiler (ekleri de dahil olmak üzere) gizlidir. Teletek Telekomünikasyon Hizmetleri A.Ş. onayi olmaksizin içerigi kopyalanamaz, üçüncü kişilere açiklanamaz veya iletilemez . Bu mesajin gönderilmek istendigi kişi degilseniz (ya da bu e-posta'yi yanlişlikla aldiysaniz), lütfen yollayan kişiyi haberdar ediniz ve mesaji sisteminizden derhal siliniz. Teletek bu mesajin içerdigi bilgilerin dogrulugu veya eksiksiz oldugu konusunda bir garanti vermemektedir. Bu nedenle bilgilerin ne şekilde olursa olsun içeriginden, iletilmesinden, alinmasindan, saklanmasindan sorumlu degildir. Bu mesajin içerigi yazarina ait olup, Teletek'in görüşlerini içermeyebilir.Disclaimer: This e-mail (including any attachments) may contain confidential and/or privileged information. Copying, disclosure or distribution of the material in this e-mail without Teletek Telekomünikasyon Hizmetleri A.Ş.'s authority is strictly forbidden. If you are not the intended recipient (or have received this e-mail in error), please notify the sender and delete it from your system immediately. Teletek makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the information transmission, reception, storage or use of such in any way whatsoever. Any opinions expressed in this message are those of the author and may not necessarily reflect the opinions of Teletek.

1 0

[Serusers] xcap module compilation issues
by Parminder.Nayyar＠relianceada.com 01 Aug '07

01 Aug '07

Hi Can anyone help me out in compiling xcap module?? as far as the module documentation concerns... its says.. that libcurl.so should be there already.. but how do i get that i downloaded a curl.tar.gz from internet.. gunzip n untar it.... and then compiled it but can't find any .so made out of it more so, the xcap module when compiled.. gives error that it can't find curl/curl.h What do i do?? Can any one help??? Parminder Nayyar

2 1

[OpenSER-Users] Problem with "log_fmt"
by zze-WU Yin RD-MAPS-LAN 01 Aug '07

01 Aug '07

Hi, everybody I use openser and want to use acc module. But when I tryed to use modaparm("acc", "log_fmt", "mfs") openser can not start. Can you tell me why?

3 2

[Serusers] How to register call forward(busy, no answer, all) from the UserAgent?
by Sean 01 Aug '07

01 Aug '07

Greetings, I would like to register the call forward functions for my UA, is it available for registering these functions from UserAgent? for example, via the REGISTER sip message or other messages? can anyone help me for this topic? Thanks a lot. Sean ____________________________________________________________________________________ Pinpoint customers who are looking for what you sell. http://searchmarketing.yahoo.com/

1 0

[OpenSER-Users] Can't use variables as function parameter?
by Iñaki Baz Castillo 01 Aug '07

01 Aug '07

Hi, in URI as "ping(a)domain.org" I need to change URI to just "domain.org" (without the username@). But I have multidomain so can't rewrite uri directly and must use the requested URI domain ($rd). So I try: if (uri=~"sip:ping[@]+.*") { rewriteuri("sip:$rd"); xlog("L_INFO", "-- New URI = $ru\r\n"); ... but I get: -- New URI = sip:$rd so it seems that I cannot use variables for making function parameters, is it true? so then, how should I do it? maybe I need using AVP's for that? Thanks for all. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

2 2

[OpenSER-Users] Info about "TM/473 Filtered"?
by Iñaki Baz Castillo 31 Jul '07

31 Jul '07

Hi, I'd like to understand how TM module filters "sometimes" some calls. For now, I just think it there isn't response and the call is repeated sometimes then TM module filteres the following calls. I find no info about how it works in the doc, could somebody give me a link or so to a document with the explanation of this issue? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

4 11

[OpenSER-Users] Colored logs in a file
by Iñaki Baz Castillo 31 Jul '07

31 Jul '07

Hi, I use colors for the XLOGs with log_stderror=yes, this is nice when I'm learning and debuggoing all the time. But for a production system I know I need to disable log_stderror and log into Syslog (for example using a specific local facility for logging in an independent file /var/log/openser.log). But in that way I loose my cool colors that help me a lot debugging. A solution I'm thinking about is to use "log_stderror=yes" and start OpenSer redirecting the output to a file, but sincerely I don't think this is very effective and serious. Any suggestion? maybe I should forget my funny colors? Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

3 6

[OpenSER-Users] Privider SIP/PSTN
by Marc LEURENT 31 Jul '07

31 Jul '07

I have a SIP/PSTN provider with an account username/password. The address of the provider is sip.test.com I would like to redirect phone calls to the gateway, but how is it possible to log in on the gateway and forward calls? Thank you for your replies Best Regards, I've tried: if (uri=~"0[0-9][0-9]*@*") { rewritehostport("username:password@sip.test.com"); # Rewrite the URI route(1); exit; }

4 10

[OpenSER-Users] Parse error in "avpops" module
by Iñaki Baz Castillo 31 Jul '07

31 Jul '07

Hi, using OpenSer 1.2.1 and trying to use "usr_preferences" MySQL table for loading AVP's. My conf: loadmodule "avpops.so" ... modparam("avpops", "db_url","mysql://openser:openserrw@localhost/openser") modparam("avpops", "avp_table","usr_preferences") modparam("avpops","use_domain",1) ... route[5] # Auth { line #543: if (avp_db_load("$ru/domain","s:trusted_ips") && avp_check("s:trusted_ips", "eq/$sp/gi")) { xlog("L_INFO", "Authorized by usr_preferences AVPs\r\n"); } .... } I get this error: 0(0) ERROR:avops:parse_avp_db:: error - bad avp flags 0(0) ERROR:avpops:fixup_db_avp: parse failed 0(0) ERROR: fix_actions: fixing failed (code=-1) at cfg line 543 0(0) ERROR: fix_expr : fix_actions error ERROR: error -1 while trying to fix configuration Any idea of what I'm doing wrong? Thanks a lot for any help. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

3 4

[OpenSER-Users] nathelper without rtpproxy
by Nhadie Ramos 31 Jul '07

31 Jul '07

Hi All, Has anyone tried nathelper but not using rtpproxy, my server is on a location with limited bandwidth so i'd like to be able to just have the sip messages to the server but not the media. Do i need all phones configured to use STUN if the SIP server is configured that way? Thank You in Advanced Regards, Nhadie

4 7

[Serusers] nathelper without rtpproxy
by Nhadie Ramos 31 Jul '07

31 Jul '07

Hi All, Has anyone tried nathelper but not using rtpproxy, my server is on a location with limited bandwidth so i'd like to be able to just have the sip messages to the server but not the media. Do i need all phones configured to use STUN if the SIP server is configured that way? Thank You in Advanced Regards, Nhadie

1 0

[OpenSER-Users] OpenSER + Freeradius + CDRTool
by Marc LEURENT 30 Jul '07

30 Jul '07

Hello, I'm still trying to get openser + freeradius + cdrtool working together I have stil one problem, it's that UserName are not stored in the acct database, whereas they are displayed in the aact logs Any piece of advice will be welcome Thanks This is a view of the CDRTool interface: SIP Caller, In, SIP destination, don't appear in the database! Id Start time SIP Proxy SIP caller In SIP destination Out Dur Price KBIn KBOut Status Codec 3N 2007-07-30 10:00:19 05:22 0.00 0.00 Ok (200) Signalling information Session: 1ec37-c0a80101-0-2(a)192.168.95.4 Start time: 2007-07-30 10:00:19 Europe/Paris Stop time: 2007-07-30 10:05:41 Europe/Paris Method: INVITE from : From: Dialed URI: Next hop URI: Billing Party: This is a part of the acct log Mon Jul 30 13:55:19 2007 Acct-Status-Type = Start Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = INVITE Event-Timestamp = "Jul 30 2007 13:55:19 CEST" Sip-From-Tag = "c0a80101-6e0646" Sip-To-Tag = "c0a80101-888e30" Acct-Session-Id = "3f9aaaf-c0a80101-0-10(a)192.168.95.4" Sip-Src-IP = "192.168.95.4" Sip-Src-Port = "5060" User-Agent = "THOMSON ST2030 hw0 fw1.50 00-0E-50-4E-AF-AE" User-Name = "105" NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 192.168.95.248 Client-IP-Address = 192.168.95.248 Acct-Unique-Session-Id = "694a4cf65dd7fd9f" Timestamp = 1185796519

2 2

[Serusers] Bad Config error in SER.CFG when tried with SEMS
by Ranganath B 30 Jul '07

30 Jul '07

Hi, When I was trying to bringup SER with SEMS installed I get the following error in /var/log/messages: Jul 30 16:24:21 Emailclient ser: parse error (214,17-18): parse error Jul 30 16:24:21 Emailclient ser: parse error (214,19-20): bad arguments The line number 214 refers to "load_avp("ruri","email",0);" Please provide me some input on getting rid of this issue. Thanks in advance, Ranganath B

1 0

[OpenSER-Users] Mediaproxy performance
by Tung Tran 30 Jul '07

30 Jul '07

5 11

[Serusers] RTP capacity
by Tung Tran 29 Jul '07

29 Jul '07

1 0

[Serusers] consultant needed
by Theodore Cekan 29 Jul '07

29 Jul '07

Hello, We are looking to hire a consultant to help us design and configure Ser for a residential service offering. We are based in Denver Colorado. If you know of anyone who could help please let me know. Ted Cekan 303-692-9409 x304

1 0

[Users] provisional_reply_timer
by Klaus Darilion 28 Jul '07

28 Jul '07

Hi! i wonder if there is a timer which controls the timeout for provisional replies. E.g. I want to cancel an INVITE and retry another gateway if the gateway does not respond with 100 (or 180) in a certain time. regards klaus -- Klaus Darilion nic.at

5 12

[OpenSER-Users] Unable to configure LCR in openser - problem with next_gw()?
by Miodrag Radulovic 28 Jul '07

28 Jul '07

Hello everyone! Please help... I've started testing openser 1.2.1 few weeks ago (on freebsd 6.2, I've got it from freebsd port collection) and I simple can not make LCR work at all. I spent last few hours reading almost all the posts on this mailing list and I can see it is working but I can not figure it out why mine configuration is not... For simple start - I want all calls where destination number starts with 9 to be routed to PSTN GW. So I made these entries in LCR tables: [root@santos /var/log]# openserctl lcr show lcr routes +----+--------+----------+--------+----------+ | id | prefix | from_uri | grp_id | priority | +----+--------+----------+--------+----------+ | 10 | 9 | % | 1 | 1 | +----+--------+----------+--------+----------+ lcr gateway groups +--------+----------+ | grp_id | grp_name | +--------+----------+ | 1 | comutel | +--------+----------+ lcr gateways +---------+---------------+------+------------+-----------+--------+-------+--------+ | gw_name | ip_addr | port | uri_scheme | transport | grp_id | strip | prefix | +---------+---------------+------+------------+-----------+--------+-------+--------+ | c2811 | 192.168.253.1 | 5060 | 1 | 1 | 1 | 0 | | +---------+---------------+------+------------+-----------+--------+-------+--------+ For openser.cfg itself I've started with sip:wizard configuration template (SIP Proxy/Registrar with Offnet-Termination and Accounting) then I made some necessary changes (I do not have Mediaproxy or RTPproxy and added domain and populated trusted table). But when I dial 93016576 I always have this error message in xlog output: Jul 19 18:50:24 santos /usr/local/sbin/openser[1412]: No PSTN gateways available - M=INVITE RURI=sip:93016576@192.168.253.205:5060 F=sip:lvs1@192.168.253.205:5060 T=sip:93016576@192.168.253.205:5060 IP=192.168.253.202 ID=676c7a05-4bcd2745(a)192.168.253.202 Here is part of openser.cfg where next_gw() fails: route[10] { if(isflagset(20)) { xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(6); exit; } route(12); if(!isflagset(23)) { # don't allow calls relaying from PSTN to PSTN, if not explicitely forwarded if(uri =~ "^sip:[0-9]+@") { # only route numeric users to PSTN if(!load_gws()) { xlog("L_ERR", "Error loading PSTN gateways - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } if(!next_gw()) { xlog("L_ERR", "No PSTN gateways available - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } setflag(21); t_on_failure("1"); route(6); } } xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(3); sl_send_reply("404", "User Not Found"); exit; } I believe that I've made some errors in LCR routes entries, but documentation is really sparse on this issue... Please note - I've tried replacing "%" with empty string in from_uri column in routes table but then openser was not able to startup: [root@santos /var/log]# openserctl lcr rmroute 9 % 1 1 MySql password for user 'openser@localhost': 200 OK [root@santos /var/log]# openserctl lcr addroute 9 '' 1 1 MySql password for user 'openser@localhost': 200 OK [root@santos /var/log]# openserctl lcr show lcr routes +----+--------+----------+--------+----------+ | id | prefix | from_uri | grp_id | priority | +----+--------+----------+--------+----------+ | 11 | 9 | | 1 | 1 | +----+--------+----------+--------+----------+ lcr gateway groups +--------+----------+ | grp_id | grp_name | +--------+----------+ | 1 | comutel | +--------+----------+ lcr gateways +---------+---------------+------+------------+-----------+--------+-------+--------+ | gw_name | ip_addr | port | uri_scheme | transport | grp_id | strip | prefix | +---------+---------------+------+------------+-----------+--------+-------+--------+ | c2811 | 192.168.253.1 | 5060 | 1 | 1 | 1 | 0 | | +---------+---------------+------+------------+-----------+--------+-------+--------+ [root@santos /var/log]# openserctl restart Stopping OpenSER : \E[37;33mINFO: stopped Starting OpenSER : \E[37;31mERROR: PID file /var/run/openser.pid does not exist -- OpenSER start failed And here is relevant part from openser log output: Jul 21 21:30:26 santos /usr/local/sbin/openser[10216]: ERROR:lcr:load_regex: bad from_uri re Jul 21 21:30:26 santos /usr/local/sbin/openser[10216]: ERROR: lcr:mod_init(): failed to reload gateways and routes Jul 21 21:30:26 santos /usr/local/sbin/openser[10216]: init_mod(): Error while initializing module lcr Please can someone help and point me in the right direction? BR, Miodrag

4 5

[Serusers] Is there something weird about this INVITE ?
by SIP 27 Jul '07

27 Jul '07

One of our clients, using a Nokia E90, is having problems connecting to anything. His INVITE looks like this: U 123.236.98.24:55031 -> 11.22.33.44:5060 INVITE sip:1747XXXXXXX@our.proxy.com;user=phone SIP/2.0. Route: <sip:our.proxy.com;lr;transport=UDP>. Via: SIP/2.0/UDP 192.168.1.229:5060;branch=z9hG4bK3e89pocn1m7mkuiefdjljsj;rport. From: <sip:1101XXXXXXX@our.proxy.com>;tag=ls57mprkkdhc7kku07de. To: <sip:1747XXXXXXX@our.proxy.com;user=phone>. Contact: <sip:1101XXXXXXX@123.236.98.24:55045;transport=UDP>. Supported: 100rel,sec-agree. CSeq: 980 INVITE. Call-ID: pyBXpNNeoIcFBM7qVqVKrBneC2kxeM. Allow: INVITE,ACK,BYE,CANCEL,REFER,NOTIFY,OPTIONS,PRACK. Expires: 120. Privacy: none. User-Agent: Nokia RA-6 V 07.24.0.3. P-Preferred-Identity: sip:1101XXXXXXX@our.proxy.com. Max-Forwards: 70. Proxy-Authorization: Digest qop=auth,realm="our.proxy.com",nonce="46a713c0c887788778a731de0355c04db1896d23",algorithm=MD5,username="1101XXXXXXX",cnonce="0dad64affaabb2fa9c93611ec864e363",nc=00000001,uri="sip:1747XXXXXXX@our.proxy.com;user=phone",response="197f8a8b4b3d39281a147c2d9d2b6a28". Content-Type: application/sdp. Accept: application/sdp. Content-Length: 447. . v=0. o=Nokia-SIPUA 63353630343979125 63353630343979125 IN IP4 123.236.98.24. s=-. c=IN IP4 123.236.98.24. t=0 0. m=audio 49152 RTP/AVP 96 0 8 97 18 98 13. a=sendrecv. a=ptime:20. a=maxptime:200. a=fmtp:96 mode-change-neighbor=1. a=fmtp:18 annexb=no. a=fmtp:98 0-15. a=rtpmap:96 AMR/8000/1. a=rtpmap:0 PCMU/8000/1. a=rtpmap:8 PCMA/8000/1. a=rtpmap:97 iLBC/8000/1. a=rtpmap:18 G729/8000/1. a=rtpmap:98 telephone-event/8000/1. a=rtpmap:13 The server received this packet, but never responds to it. It simply never sends a reply. It's as though it doesn't recognise this as a proper INVITE. The ONLY thing I noticed about this that's completely non-standard, is that every other packet seems to have this period after it (indicating, perhaps, the end of a line... I don't know enough about the display of ngrep to tell). This INVITE does not. It goes almost immediately into the next packet, and it's almost as though this one never ends. Has anyone seen this before? Is there anything weird about this INVITE that would cause SER to ignore it? N.

3 2

[OpenSER-Users] Re: [Serusers] CDRz?
by Parminder.Nayyar＠relianceada.com 27 Jul '07

27 Jul '07

Hello Friends ... Jai, Thanks for showing a pointer towards teh acc module... but i really cudn't grasp even a bit out of it... all it says is about logs being dumped, with specific data like \: The fixed minimal accouting information is: Request Method name >From header TAG parameter To header TAG parameter Call-Id 3-digit Status code from final reply Reason phrase from final reply Time stamp when transaction was completed There is no session/dialog accounting (yet)-- OpenSER maintains no sessions. If one needs to correlate INVITEs with BYEs for generating proper CDRs for example for purpose of billing, then it is better done in the entity which collects accounting information. If a UA fails in middle of conversation, a proxy will never learn it. In general, a better practice is to account from an end-device (such as PSTN gateway), which best knows about call status (including media status and PSTN status in case of the gateway). Now, my requirement is to generate a CDR for billing purpose... In syslog there are not any logs present, and even if i get the logs there.. using changes in config file, how do i use this raw information to generated CDRz for billing purposes??? I have searched for quite a few tools online which work on the output of SER to generate CDR or for CDR collection... but I need a free software, they are all paid services... Can someone help me where should i go further..? I'm lost ... :( Thanks a lot! Parminder Nayyar "Jai Rangi" <jprangi(a)gmail.com> 07/23/2007 10:52 PM To "Parminder.Nayyar(a)relianceada.com" <Parminder.Nayyar(a)relianceada.com> cc Subject Re: [Serusers] CDRz? acc module can be used to all CDRs. But I found that useless for billing purpose. Jai www.bingotelecom.com On 7/22/07, Parminder.Nayyar(a)relianceada.com < Parminder.Nayyar(a)relianceada.com> wrote: Hi Can anyone tell me which module to explore , if i want to check about CDRz and billing?? Please help..... Parminder Nayyar _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

3 5

[Serusers] can SIP client connect sip proxy via SOCKS5 proxy?
by Liu Wenlong 27 Jul '07

27 Jul '07

Is there any protocol support it? thx -- aRES iN aNYWhere

1 0

[OpenSER-Users] t_on_branch -- using drop() question?
by Zahid Mehmood 26 Jul '07

26 Jul '07

I was looking at the openser config generated using the online sipwise wizard. They implement conditional call forwarding using: if($avp(s:cfc) != NULL) { avp_pushto("$ru", "$avp(s:cfc)"); setflag(29); append_branch(); t_on_branch("1"); xlog("L_INFO", "CFC detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(13); } branch_route[1] { if(is_domain_local("$rd")) { xlog("L_INFO", "Dropping local branch - M=$rm RURI= $ru F=$fu T=$tu IP=$si ID=$ci\n"); drop(); } } Suppose an invite gets to this point and ruri is changed to another local user. does this drop() in branch_route prevents that invite to be relayed? does it still go through route(13) ? I'm still a newbie trying to better understand the working of openser functions. I'll greatly appreciate if someone can briefly describe when it makes sense to use "append_branch()" Thanks in advance for your help. -- Zahid

2 3

[OpenSER-Users] openSER crashes while accessing emtpy pseudo-variable $fn in if-clause ?
by Rainer Strobel 26 Jul '07

26 Jul '07

Hello, while playing around with the pseudo variables whithin openser 1.2.1 I got a strange behavior. I tried to check if the from-header-name of an invite message is set to (A|a)nonymous with an simple if-clause. if($fn =~"(A|a)nonymous") { ...do something ; } It works fine while the from-name variable is not empty. If the from-header-name in the INVITE message is not given openser crashes exactly on that comparison. After that I started playing around using other comparisons, for example 'if($fn=~"foo")' or ' if($fn == "foo")' , with the same set-up. Openser crashed each time processing that if-clause. I tested this with openser-1.2.1-notls on i386 openSUSE 10.2 with MySQL 5.0.26 and gcc 4.1.2 Can anybody confirm this problem ? Debuggin informations are available and can be send additional if necessary. Regards. Rainer Strobel --------------------------------------------------------------------- HFO Telecom AG Network Management Bahnhofstraße 18 95028 Hof Tel.: + 49 (0) 92 81 / 14 48 132 Fax: + 49 (0) 92 81 / 14 48 52 132 E-Mail: strobel(a)hfo-telecom.de Internet: www.hfo-telecom.de

2 1

[OpenSER-Users] Problem with handling 3xx responses and contact with maddr
by Watkins, Bradley 26 Jul '07

26 Jul '07

I'm using OpenSER to connect to a Nortel CS1000M, and I'm having some problems. The way the Nortel works is that it has what they call a Network Routing Service (redirect server, really), that issues either a 302 (if a single SIP gateway exists on the CS1000) or a 300 (if multiple). Here's a sample 300 from my system (IPs and domains changed to protect the guilty): U 10.1.1.3:5060 -> 10.0.1.1:5060 SIP/2.0 300 Multiple Choices. Via: SIP/2.0/UDP 10.0.1.1;branch=z9hG4bKa66e.99fc1a25.1;received=10.0.1.1. Via: SIP/2.0/UDP 10.1.3.114;branch=z9hG4bK7d42804867A25509. From: "0004f2145263" <sip:0004f2145263@dtw-openser01.sub.domain.com>;tag=5F7CDD45-DDCE1F92. To: <sip:12345@dtw-openser01.sub.domain.com;user=phone>;tag=34887. Call-ID: 58dd5646-e86d140f-c2ead634(a)10.1.3.114. CSeq: 1 INVITE. Contact: <sip:12345;phone-context=udp@domain.com:5060;maddr=10.1.1.1;transport=ud p;x-nt-redirect=redirect-server;x-nt-redirect=redirect-server>. Contact: <sip:12345;phone-context=udp@domain.com:5060;maddr=10.1.1.2;transport=tc p;x-nt-redirect=redirect-server;x-nt-redirect=redirect-server>. Content-Length: 0. The problem I have is that the Contacts have an maddr, and I need relay to it. However, I can't seem to be able to use transformations to get the maddr for relaying. Here are some log messages with comments inline: Jul 16 19:50:49 dtw-openser01 /usr/local/sbin/openser[4306]: Branches are: <sip:12345;phone-context=udp@domain.com:5060;maddr=10.1.1.1;transport=ud p;x-nt-redirect=redirect-server>;q=0.01 This is just an xlog of $bR, but I'm not sure where the q= value is coming from (get_redirects?). Jul 16 19:50:49 dtw-openser01 /usr/local/sbin/openser[4306]: ERROR: parse_uri: bad uri, state 0 parsed: <<sip> (4) / <<sip:12345;phone-context=udp@domain.com:5060;maddr=10.1.1.1;transport=u dp;x-nt-redirect=redirect-server>;q=0.01> (120) Jul 16 19:50:49 dtw-openser01 /usr/local/sbin/openser[4306]: tr_eval_uri: invalid uri [<sip:12345;phone-context=udp@domain.com:5060;maddr=10.1.1.1;transport=u dp;x-nt-redirect=redirect-server>;q=0.01] These are what I get as a result of trying to xlog $(bR{uri.maddr}) Anyway, I would be greatly appreciative of any help someone might offer on the matter. Regards, - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.

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[OpenSER-Users] [multidomain] Using different RtpProxy for each domain
by Iñaki Baz Castillo 26 Jul '07

26 Jul '07

Hi, I've configured a OpenSer in multidomain with a RtpProxy. Now I'd like to use a RtpProxy for each domain, so it could be based in $rd. But at first point I don't know how to configure OpenSer to use varios RtpProxies in different machines. I've tried with: modparam("nathelper", "rtpproxy_sock", "udp:localhost:22222,udp:192.168.1.58:22222") and: modparam("nathelper", "rtpproxy_sock", "udp:localhost:22222udp:192.168.1.58:22222") but it fails when starting giving an error of parsing. If I use: modparam("nathelper", "rtpproxy_sock", "udp:localhost:22222,udp") modparam("nathelper", "rtpproxy_sock", "udp:192.168.1.58:22222") just the last line is used. Any idea of how to use various RtpProxies? My idea is to have some of them availables and use them based in $rd using: force_rtp_proxy("","192.168.1.58") and so. Any help please? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] What does "script writer didn't release transaction" mean?
by isetien001＠ikasle.ehu.es 26 Jul '07

26 Jul '07

Hi! I get the following warning when I start openser. Any idea of what it is? I'm using Openser with the xmpp module. Thanks, Izortze. # openser start Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 158.227.64.88 [158.227.64.88]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 158.227.64.88 [158.227.64.88]:5060 Aliases: tcp: b001948.bi.ehu.es:5060 tcp: localhost:5060 udp: b001948.bi.ehu.es:5060 udp: localhost:5060 0(21183) init_tcp: using epoll_lt as the io watch method (auto detected) 0(0) INFO: statistics manager successfully initialized 0(0) StateLess module - initializing 0(0) TM - initializing... b001948:/usr/local/etc/openser# 0(0) Maxfwd module- initializing 0(0) INFO:ul_init_locks: locks array size 512 0(0) TextOPS - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 107520 0(0) INFO: udp_init: SO_RCVBUF is finally 215040 0(0) INFO: udp_init: SO_RCVBUF is initially 107520 0(0) INFO: udp_init: SO_RCVBUF is finally 215040 1(21186) INFO:mi_fifo:mi_child_init(1): extra fifo listener processes created 5(21190) WARNING: script writer didn't release transaction 6(21191) WARNING: script writer didn't release transaction 7(21192) WARNING: script writer didn't release transaction 6(21191) WARNING: script writer didn't release transaction 8(21193) WARNING: script writer didn't release transaction 7(21192) WARNING: script writer didn't release transaction 8(21193) WARNING: script writer didn't release transaction 5(21190) WARNING: script writer didn't release transaction 7(21192) WARNING: script writer didn't release transaction 5(21190) WARNING: script writer didn't release transaction 7(21192) WARNING: script writer didn't release transaction

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[Serusers] ser.cfg syntax error
by Emiliano Ogando 26 Jul '07

26 Jul '07

This is the beginning of my ser.cfg file I've got syntax error on line 12, when I run the ser process. The ser_fifo file should be created when I compile ser or when I run this script?? debug=3 fork=no log_stderror=yes listen=192.9.208.66 port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" #line 12 Regards Emiliano Ogando Dirección General de Gestión Informática Ministerio del Interior 25 de Mayo #101 TE: (54-11) 4346-1663 Fax: (54-11) 4346-1633 Web: <http://www.mininterior.gov.ar/> http://www.mininterior.gov.ar Buenos Aires - República Argentina Nota: todos los mails enviados desde esta cuenta eogando(a)mininterior.gov.ar<mailto:eogando@mininterior.gov.ar> serán válidos únicamente si se encuentran firmados digitalmente, certificados por la autoridad Oficina Nacional de Tecnologías de la Información, de la Jefatura de Gabinete de Ministros de la Presidencia de la Nación Argentina.

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[OpenSER-Users] Regarding parse error
by Ashish Barot 26 Jul '07

26 Jul '07

Hello all, right now i had just try to install openser-1.2.1-notls it get installed with out any error while doing make and make install but after that if i try to start service then it is giving me error openser: parse error (23,1-5): syntax error Jul 27 04:46:39 barot1 openser: parse error (23,1-5): i get above error when i try to run "openser start" command. i am using Fedora Core-6 as my OS. can anybody guide me on this? Thanks. Best Regads, Ashish Barot.

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[OpenSER-Users] Concept Question Regaring OpenSer and MediaProxy
by Christopher Aloi 25 Jul '07

25 Jul '07

Hello All - First - I am new to SER; but versed in telephony, so apologies for any incorrect verbiage. I've been reading a bit on OpenSer and MedaProxy. I plan to build an proxy to connect two separate segments of our internal network. I will need SER to pass signaling and media from networkA to networkB. There will be no NAT. My question is -> All the reading i've done regarding proxying media with OpeSer relates to NAT traversal solutions; do the core concepts still apply for traversing multiple networks w/no NAT? Thank, Chris

1 0

[OpenSER-Users] Language encoding
by Greg Fausak 25 Jul '07

25 Jul '07

I hope someone can help me with a language encoding problem. I don't have much experience with non-ascii From headers. What is the correct way to encode in the SIP headers? The reason I ask is because we're having problems with parse_from() barfing on 8859-1 characters. Do we need to just use utf-8? Are other charsets supported? Thanks, -- Greg Fausak greg(a)thursday.com

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[Serusers] FW: SER Fail Over
by Ravin Suri 25 Jul '07

25 Jul '07

Can you elaborate what is "heartbeat"? Thanks for your help. - Ravin _____ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Jai Rangi Sent: Tuesday, July 24, 2007 9:30 PM To: Arun Kumar Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] SER Fail Over Arun, You can use heartbeat. That way you will be able to use the same IP for both ser servers. Thats what we have in our callingcard application. -Jai www.bingotelecom.com <http://www.bingotelecom.com> On 7/24/07, Arun Kumar <arunvoip(a)gmail.com> wrote: Hi I've two sites running my ser with two different database (master-slave). How do I configure fail over between my ser ? thanks arun _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[OpenSER-Users] Wrong/Double text appears in Contact header during sequential forking
by liran tal 25 Jul '07

25 Jul '07

Hey everyone, I'm using sequential forking and on one of the scenarios there appears to be a problem. When OpenSER attempts to find the first most relevant destination for the call the SIP headers are ok. If the first destination that OpenSER attempts to contact is offline/unreachable it continues to the next one in turn in which it produces a bad Contact header which looks like this: Contact: <sip:101@192.168.0.1:5060sip:101@192.168.0.1:5060> As you can see it's writing the sip information twice for some reason. Has anyone seen this happen before? Also, where should I be looking at to find the problem? Thanks, Lir.

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[OpenSER-Users] OpeSER Aministrator Training Course at VoN Itally, Rome - Details
by Bogdan-Andrei Iancu 25 Jul '07

25 Jul '07

Hi everybody, As promised, we finalized all the arrangements and here are the last details about the "OpenSER Admin Training Course" at VoN Italy, Rome. Prices, dates, agenda, registration forms and more others can be found on the OpenSER project site, at: http://www.openser.org/mos/view/OpenSER-Admin-Course---Rome-2007/ The registration process is on, so feel free and join :). Thanks and regards, Bogdan

2 3

[Serusers] SER-routing-problem
by Vlad Costea 25 Jul '07

25 Jul '07

Hello. I have just installed SER on RedHat 9 Linux with the purpose to route voice packages between a PSTN Gateway AS5350 and a Grandstream HandyTone 286.Letme tell you how it should work: I have assigned a phone number, that when it is picked-up, the IVR from the AS5350 router respods and after the press of a key it sends the packages to the HandyTone and from there to a normal phone; So far so good because it partialy works, that means that the phone rings butt there is no voice. I have managed to solve this problem using a hardware Sip Server between the two devices ( in this way both HT286 and AS5350 act as clients), but this is not possible any more because it was not my server; so i have tried with a software solution and I've installed SER. As I said before , I haven't managed to configure it to work as I wish and this is why I'am asking for your help; it would realy help me if you can provide a ser.cfg example that would do just the routing part (no ack no authentification , no mysql,...., just routing). Please excuse my bad english. Thank you very much !

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[OpenSER-Users] Info about OpenSer Von Italy course
by Iñaki Baz Castillo 25 Jul '07

25 Jul '07

Hi, I'm interested in going to that course but I get no info about it. I sent some mails with no response. I'd just need to know if I could buy the plane tickets and reserve hotel now or I should wait for the registration process. Could somebody give me some info about it? Thanks for all and regards. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

2 2

[OpenSER-Users] linux kernel + iptables + sip nat module + RTP streams
by Marc LEURENT 25 Jul '07

25 Jul '07

Good Morning, I need to set a VOIP architecture with natted users. And I'm trying to prevent bottleneck (with a rtpproxy architecture) and I have a full access to the nat server (debian box / iptables MASQUERADE NAT). So, I've compiled nf_nat_sip / nf_conntrack_sip modules that changed the received RTP stream IP with the public one of my nat box. So the RTP stream is arriving to the natbox! Do you have any idea of a solution to forward RTP streams to the end user Best Regards, Marc

1 0

[OpenSER-Users] Problem matching GTALK with Openser
by isetien001＠ikasle.ehu.es 25 Jul '07

25 Jul '07

Hi! I have load the XMPP module to my openser configuration to integrate GTALK in my SIP system designed with Openser. The XMPP module works in the 'component' mode. I have added the server Jabberd 2 to my system and I have checked that it works correctly with different Jabber clients. When I start Openser and try to connect GTALK to it I get the following error: 10(8016) ERROR:parse_first_line: error parsing the method 10(8016) ERROR:parse_first_line: bad message 10(8016) ERROR: parse_msg: message=<CONNECT www.google.com:443 HTTP/1.0 User-Agent: Google Talk Host: www.google.com Content-Length: 0 Proxy-Connection: Keep-Alive > 10(8016) ERROR: receive_msg: parse_msg failed 10(8016) ERROR: receive_msg: no via found in request 10(8016) ERROR: receive_msg: no via found in request Any idea of what it's wrong? Thanks in advance! Izortze. P.D= This is my openser.cfg file: $Id: openser.cfg 1676 2007-02-21 13:16:34Z 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 # check_via=no rev_dns=no dns=no port=5060 #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 = "/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" loadmodule "xmpp.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "auth.so" #loadmodule "auth_db.so" # ----------------- setting module-specific parameters --------------- # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) #-- xmpp module parameters -- modparam("xmpp","backend","component") modparam("xmpp","domain_separator","*") modparam("xmpp","gateway_domain","b001948.bi.ehu.es") modparam("xmpp","xmpp_domain","b001948.bi.ehu.es") modparam("xmpp","xmpp_password","secret") modparam("xmpp","xmpp_host","b001948.bi.ehu.es") # ------------------------- 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; }; # filter XMPP destinations if (uri =~ "sip:.+\*b001948.bi.ehu.es@b001948.bi.ehu.es") { # absorb retransmissions if (!t_newtran()) { sl_reply_error(); exit; } # handle IM if (is_method("MESSAGE")) { log("message to XMPP: from <$fu> to <$ru>\n"); if (xmpp_send_message()) { sl_send_reply("202", "Accepted"); } else { sl_send_reply("404", "Not found"); } exit; } # unsupported type of SIP message log("message to XMPP: method <$rm> not supported yet\n"); sl_send_reply("503", "Service unavailable"); exit; } else { # 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("openser.org", "subscriber")) { # www_challenge("openser.org", "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; }

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[OpenSER-Users] Multidomain support
by Iñaki Baz Castillo 25 Jul '07

25 Jul '07

Hi, I want to mantain independent domains in OpenSer. In my case I've a OpenSer with a single DNS A record and various CNAME (I still don't want to play with SRV and so). so: DNS A = openser.domain.org CNAME = sip1.domain.org CNAME = sip2.domain.org And I want users of sip1.domain.org and sip2.domain.org, as independent groups. I just want to avoid SIP interdomain messages, so 200(a)sip1.domain.org CAN'T invite 300(a)sip2.domain.org even if he does authentication. I've loaded "domain" module and use "is_uri_host_local()" and "is_from_local()" functions, but for now I only used one domain. My question is very general: for implement (or avoid) interdomain comunication, do I need to use the "domainpolicy" [1] module? I've read its doc and know it's based in 3 drafts [2][3][4], but all of them seems to be based in the complex NAPTR record and so. Is it the way? I think I could just compare the FROM domain with the TO domain in order to avoid interdomain communication, but of course I'd like in the future the possiblity of allowing some domains to contact some other domains. Is then "domainpolicy" the solution I should learn? Any other doc about it? Thanks for any help. [1] http://www.openser.org/docs/modules/1.2.x/domainpolicy.html [2] http://tools.ietf.org/id/draft-lendl-domain-policy-ddds-02.txt [3] http://tools.ietf.org/id/draft-lendl-speermint-federations-02.txt [4] http://tools.ietf.org/html/draft-lendl-speermint-technical-policy-00 -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] Is not possible to register in OpenSer dokuwiki?
by Iñaki Baz Castillo 25 Jul '07

25 Jul '07

I just see the option to login but not register, is not possible? Anyway it allows anonymous edit. PD: This is because I see an error in: http://www.openser.org/dokuwiki/doku.php/pseudovariables:1.2.x#domain_of_de… There was: Domain of destination URI $dd - reference to domain of destination uri but that is not correct at least in 1.2.1 version. I've changed that to $fd. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] "openserctl online" command showing user@domain
by Iñaki Baz Castillo 25 Jul '07

25 Jul '07

Hi, I've configured a multidomain OpenSer but "openserctl online" just shows the usernames and not the domain. Is it possible to get a list of registered user with syntax "user@domain"? Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

2 2

[OpenSER-Users] TM request function - callback data parameter
by Tiberius DULUMAN 25 Jul '07

25 Jul '07

Hello, I'm confused with the callback data parameter of request function of TM module: does TM takes the ownership of the callback data? I've seen that, if there is an error, t_request function frees the callback data. Also, I've seen empty_tmcb_list function that free that callback data, too. I've checked other modules, to see how do they use this parameter, and I've got even more confused. msilo.c : it initialize the pointer from an int; there is a problem in request function from uac.c file, after the 'err:' label, when 'if (cp) shm_free(cp)' is executed. pua.c and send_publish.c: the callback functions free the callback data parameter; if the TM took the ownership, then that memory is freed twice. So, what is the right way to handle the memory pointed by callback data? Thank you _________________________________________________________________ Don't just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/

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[OpenSER-Users] forward one user to a spezific ip
by NicklasZ 25 Jul '07

25 Jul '07

Hi what do i have to do to forward only 1 user to a spezific IP address on INVITE? if(method=="INVITE") { if (uri=~"^sip:\200" || src_ip == xxx.xxx.xxx.xxx) { ??????? it should be forwarded on fixed ip yyy.yyy.yyy.yyy:5060 } }; i think its very easy but i'm a greenhorn ;) Thanks Andy

2 1

[OpenSER-Users] regarding openser.cfg file
by Ashish Barot 25 Jul '07

25 Jul '07

Respected All, right now i had installed the openser-1.0.1 in my /usr/local/src/openser diectory, so i want to know which one is my main configuraion file, [i.e., openser.cfg] as i am finding that file at two places, and they are [1] /usr/local/etc/openser/openser.cfg [2] /usr/local/src/openser/openser-1.0.1/etc/openser.cfg Now i want to know, when we start openser, which file will be called ? As i had check both of those files, and there is no difference between them. so i want to understand that, which file is called first? Thanks. Best Regards, Ashish Barot.

2 1

[OpenSER-Users] radius_is_user_in
by Marc LEURENT 25 Jul '07

25 Jul '07

Thanks to you, my openser is working well with freeradius!! I still have a few questions: * the function radius_is_user_in("From","my_group") always returns false. modparam("group_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("group_radius", "use_domain", 1) if(is_method("INVITE")) { if (!radius_is_user_in("From","voip")) { xlog("L_DBG","NOT IN THE voip GROUP \r\n"); sl_send_reply("403", "Forbidden - Not in the voip group"); } else { xlog("L_DBG","IN THE voip GROUP\r\n"); }; }; I've monitored the mysql.log file, and no requests is sent to the sql server to know if the user is owned by the voip group.. I you have any idea, I would be very grateful

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[OpenSER-Users] openser database description auto-generated
by Daniel-Constantin Mierla 24 Jul '07

24 Jul '07

Hello, thanks to Henning Westerholt efforts to develop the new db xml schema, the description of OpenSER database for devel version is automatically generated in HTML format and uploaded at: http://www.openser.org/docs/db-tables/openser-db-devel.html The update is done on a daily base (hopefully), only when something changes in db schema. Links from dokuwiki and docs index were added. Cheers, Daniel

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[Serusers] SER Fail Over
by Arun Kumar 24 Jul '07

24 Jul '07

Hi I've two sites running my ser with two different database (master-slave). How do I configure fail over between my ser ? thanks arun

2 1

[Serusers] Configuring SER to work as SIP proxy
by dzus＠mat.uni.torun.pl 24 Jul '07

24 Jul '07

All, I am trying to configure SER to work as SIP proxy (not registar and not redirect). I tried to modify simple ser.cfg which comes with SER by adding one line in route[0]: rewritehostport("192.168.1.2:5060"); This unfortunately doesn't work. The message to my SIP server 192.168.1.2 are indeed forwarded. But message comming back from 192.168.1.2 are looped by my SIP proxy (192.168.1.3) back to SIP server (192.168.1.2). Can someone help me? Or send a working cfg for SIP proxy, I cannot find any. This is my complete ser.cfg: # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "//lib/ser/modules/mysql.so" loadmodule "//lib/ser/modules/sl.so" loadmodule "//lib/ser/modules/tm.so" loadmodule "//lib/ser/modules/rr.so" loadmodule "//lib/ser/modules/maxfwd.so" loadmodule "//lib/ser/modules/usrloc.so" loadmodule "//lib/ser/modules/registrar.so" loadmodule "//lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "//lib/ser/modules/auth.so" #loadmodule "//lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { 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 # Dzus rewritehostport("192.168.1.2:5060"); if (!t_relay()) { sl_reply_error(); }; } ---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program.

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[OpenSER-Users] Error with authentication "ha1"
by Iñaki Baz Castillo 24 Jul '07

24 Jul '07

Hi, I've configured: modparam("auth_db", "calculate_ha1", 0) modparam("auth_db", "password_column", "ha1") # Default value modparam("auth_db", "password_column_2", "ha1b") # Default value Twinkle softphone uses digest with "username=user@domain" so OpenSer verifies the hash with the "ha1b" column and it works well, it's authenticated. But Linksys PAP2 uses digest with just "username=user" so OpenSer uses "ha1" column. I don't know why but it fails. Of course both can authenticate perfectly using modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") Both users are generated with "openserctl add" command. Any idea why it fails with Linksys PAP2? I use OpenSer 1.2.1 of Debian testing. Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

1 1

[OpenSER-Users] Table Acc
by Gerson Matiolli 24 Jul '07

24 Jul '07

This is my table acc ! +----+--------+-----------------+-----------------+--------------------------------------------+----------+------------+---------------------+ | id | method | from_tag | to_tag | callid | sip_code | sip_reason | time | +----+--------+-----------------+-----------------+--------------------------------------------+----------+------------+---------------------+ | 69 | INVITE | e4bc67c9-21223 | ab954913-780382 | D9B3-9259-47021223FCDCBF2499AE-004@SipHost | 200 | OK | 2007-07-24 08:36:51 | | 70 | BYE | e4bc67c9-21223 | ab954913-780382 | D9B3-9259-47021223FCDCBF2499AE-004@SipHost | 200 | OK | 2007-07-24 08:37:26 | | 71 | INVITE | 202c3073-21266 | c9e16a50-780425 | D9B3-9259-470212668A329AEBB48A-005@SipHost | 200 | OK | 2007-07-24 08:37:34 | | 72 | BYE | 202c3073-21266 | c9e16a50-780425 | D9B3-9259-470212668A329AEBB48A-005@SipHost | 200 | OK | 2007-07-24 08:37:59 | +----+--------+-----------------+-----------------+--------------------------------------------+----------+------------+---------------------+ Where I meet the information of FROM_URI & TO_URI?

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[OpenSER-Users] NEW FEATURE: new logging design
by Bogdan-Andrei Iancu 24 Jul '07

24 Jul '07

Hi, OpenSER trunk has a new logging design, covering new functionalities (for end-user), more simplicity (for developers), better log format, etc. End-user functionalities ========================= 1) configurable debug level at runtime - so far we had static debug level, but now, there is a MI function to change it at runtime. If you want to have or not a configurable debug level, there is a compile option named CHANGEABLE_DEBUG_LEVEL (see Makefile.defs). By default is on, but, if there are performance concerns, you can turn it off. The MI function is called "debug": # display the current debug level openserctl fifo debug # set a new debug level (L_ERR) openserctl fifo debug -1 The function changes the global debug level, so all processes and all code will be affected. 2) script function for changing the debug level - this functions allows you to change the debug level from script, only for a specific script block. Only the debug level of the current process will be changed. This is very helpful if you are tracing and debugging only a specific piece of code. The script function is: setdebug(4); # set debug level for debug setdebug(); # reset the debug level to the global one Ex: debug= -1 # err only ..... { ...... setdebug(4); uac_replace_from(....); setdebug(); ....... } The function change the debug level of the current process and changing the global debug level (MI function) does not affect it, so be careful and reset the pre process debug level when you are done. More simplicity ================= The logging function used in code was replaced with a new set of functions, more simple and flexible: LOG / DBG were replaced with: LM_ALERT( ...) LM_CRIT( ...) LM_ERR( ...) LM_WARN( ...) LM_NOTICE( ...) LM_INFO( ...) LM_DBG( ...) Prototype is LM_XXX( "FMT", .....); The functions will automatically add to the logged message the log level, the module name, function name, process pid and time stamp, so you do not have to care about it from the code. Better log format ================== Even if you use syslog or stderr logging, the log message will contain information about log level, the module name, function name, process pid and time stamp. For syslog, the log level is mapped over the syslog levels, so no text will be used - you can configure syslog to use different files for each log level. Also time and pid are not pushed from openser as this info is automatically logged by syslog. For stderr logging, openser take care and appends the log level as text, time and pid (as text). Ex: Jul 23 19:52:19 [0] INFO:sl:mod_init: Initializing StateLess engine So, either you are using syslog, either stderr, you will still get the same information logged. We still need to migrate all the log messages from script from the old to the new set of functions, so, for the moment, not all logs will be "nice" formatted. Regards, Bogdan

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[OpenSER-Users] OpenSER + FreeRadius + CDRTool
by Marc LEURENT 24 Jul '07

24 Jul '07

Good Morning everybody, I hope you had a pleasant week-end! I'm trying to set up an infrastructure for prepaid calls for my school working with OpenSER + Freeradius + CDRTool OpenSER is working well with FreeRadius, but I have followed the CDRTool installation tutorial and now Freeradius doesn't work anymore... Both OpenSER and freeradius data is stored in an mysql database, but when I was installing CDRTool, I had to modify their database following http://cdrtool.ag-projects.com/doc/INSTALL.txt Is there someone who succeed in setting up this installation... Any Help or just any hope to succeed? PS: I'm using OpenSER 1.2.1-1 compiled by hand, Freeradius 1.1.3-3 and cdrtool_5.2.1_all.deb Marc

2 8

[OpenSER-Users] SipSpy release (How to see sip messages in openser, or whatever SIP software you use)
by Elias Baixas 24 Jul '07

24 Jul '07

Hi all, just wanted to let you know that we have released our visual SIP monitoring tool as OSS (GPL'd). You can take a look on what it does (and how) on http://www.wesip.com/mediawiki/index.php/SipSpy basically, SipSpy is composed of 2 softwares: an agent and a manager. You run the agent in all the nodes where you want to monitor SIP traffic, and the manager (User Interface) in your computer. THen the manager (SipSpy) connects to each of the agents through TCP, and they report any SIP packet that matches a given regexp to the manager. MOre or less its like having a couple of ngreps running on different hosts, but reporting the seen SIP packets to a nice graphical User Interface, so that you can see what's happening in all the nodes in real time from your desktop. ngrep is tough for starters ! Elias

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[Serusers] Transfering incoming call on SER to Asterisk
by inge 24 Jul '07

24 Jul '07

Hi all, Anyone know how can I transfer an incoming call from SER to an Asterisk ? The sip uri wich comes from SER is like : sip:enum@domain.tld But on Asterisk enum will not be necessary the extension. IT seems that with a single rewritehostport to Asterisk, it doesn't run. Thanks for your support Adrien

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