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37461 discussions

[Serusers] Suggestion Required
by Abdul Lateef 30 Jun '06

30 Jun '06

Hi friends, I am new in OpenSER, I want to use it only for SIP Proxying with freeradius. I made plan to install openser with freeradius on Virtual Server to get only 100 cuncurent calls. 1- Is it possible to install on Virtual Server? 2- Which Codecs or used, because i want to calculate the bandhwidth according to the codec? 3- What RAM should be used to handel 100 cuncurent calls? 4- It can accept h323-credit-time from radius to control max credit call time? I will appriciate for your kind of suggestion. Regards, __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] (no subject)
by Kavvalaki Alexandra 30 Jun '06

30 Jun '06

Hello, Can anyone tell me how can i create a new module? Thanks

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[Serusers] Assure a BYE is received ??
by Adrian Georgescu 30 Jun '06

30 Jun '06

I do now know your setup but you only need to add some extra columns to your radacct table. Except for the time required to apply the sql changes (your table will lock and you cannot write to it) I am not aware of any impact on a standard freeradius installation. Adrian >>>> Ok, it means that I'll just make the RTP pass through mediaproxy by enabling the use of it on non-NATed UA's. Just a question, I have an existing RADIUS accounting setup, would there be a problem if I implemented the accounting feature of mediaproxy? I'm afraid because written in the docs in the distro i need to patch the radacct table, will it affect the existing one or alter something? Thanks, Ryan

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[Serusers] crash info - DBG: tcp_main_loop: dead child 2 (shutting down?)
by Hakan YASTI 30 Jun '06

30 Jun '06

Dear All, I user ser-0.9.3_1. Today it has crashed and I got the following logs from debug.log. What can be the problem? Best Regards, Hakan. Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: SIP Reply (status): Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: version: <SIP/2.0> Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: status: <100> Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: reason: <Trying> Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: parse_headers: flags=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: Found param type 232, <branch> = <z9hG4bK846c.8bba40d.0>; state=16 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: end of header reached, state=5 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: parse_headers: Via found, flags=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: parse_headers: this is the first via Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: After parse_msg... Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: grep_sock_info - checking if host==us: 11==11 && [84.51.32.21] == [84.51.32.21] Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: grep_sock_info - checking if port 5060 matches port 5060 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: forward_reply: found module tm, passing reply to it Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: DEBUG: t_check: msg id=257654 global id=257653 T start=0xffffffff Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: parse_headers: flags=17 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: Found param type 232, <branch> = <z9hG4bKacHxBfRNa>; state=16 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: end of header reached, state=5 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: parse_headers: Via found, flags=17 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: parse_headers: this is the second via Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: end of header reached, state=9 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: DEBUG: get_hdr_field: <To> [43]; uri=[sip:009xxxxxxxxx@mydomain;user=phone] Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: DEBUG: to body [<sip:009xxxxxxxxxx@mydomain;user=phone>^M ] Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: get_hdr_field: cseq <CSeq>: <1> <INVITE> Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: parse_headers: flags=4 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: DEBUG: t_reply_matching: hash 50760 label 218409912 branch 0 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: DEBUG: t_reply_matching: reply matched (T=0x2877aa78)! Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: DEBUG: t_check: msg id=257654 global id=257654 T end=0x2877aa78 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: DEBUG: reply_received: org. status uas=100, uac[0]=0 local=0 is_invite=1) Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: ->>>>>>>>> T_code=100, new_code=100 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: DEBUG: relay_reply: branch=0, save=0, relay=-1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: DEBUG: add_to_tail_of_timer[1]: 0x2877aba0 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: DEBUG:destroy_avp_list: destroying list 0x0 Jun 26 14:37:56 globalser /usr/local/sbin/ser[640]: receive_msg: cleaning up Jun 26 14:37:56 globalser /usr/local/sbin/ser[666]: DBG: tcp_main_loop: dead child 2 (shutting down?) Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: Memory status (pkg): Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: qm_status (0x8103e40): Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: heap size= 1027664 Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: used= 807328, used+overhead=1018224, free=9440 Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: max used (+overhead)= 1026976 Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: dumping all alloc'ed. fragments: Jun 26 14:37:56 globalser /usr/local/sbin/ser[643]: Memory status (pkg): Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: 0. N address=0x8108fe8 frag=0x8108fe0 size=128 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[643]: qm_status (0x8103e40): Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: 1. N address=0x8109078 frag=0x8109070 size=128 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[643]: heap size= 1027664 Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: 2. N address=0x8109108 frag=0x8109100 size=32 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[643]: used= 809296, used+overhead=1020912, free=6752 Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: 3. N address=0x8109138 frag=0x8109130 size=16 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[643]: max used (+overhead)= 1028288 Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: 4. N address=0x8109158 frag=0x8109150 size=16 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[643]: dumping all alloc'ed. fragments: Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: 5. N address=0x8109178 frag=0x8109170 size=112 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[643]: 0. N address=0x8108fe8 frag=0x8108fe0 size=128 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: 6. N address=0x81091f8 frag=0x81091f0 size=16 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[643]: 1. N address=0x8109078 frag=0x8109070 size=128 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: 7. N address=0x8109218 frag=0x8109210 size=112 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[643]: 2. N address=0x8109108 frag=0x8109100 size=32 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: 8. N address=0x8109298 frag=0x8109290 size=16 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[643]: 3. N address=0x8109138 frag=0x8109130 size=16 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: 9. N address=0x81092b8 frag=0x81092b0 size=16 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[643]: 4. N address=0x8109158 frag=0x8109150 size=16 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: 10. N address=0x81092d8 frag=0x81092d0 size=16 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[643]: 5. N address=0x8109178 frag=0x8109170 size=112 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[644]: 11. N address=0x81092f8 frag=0x81092f0 size=16 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[643]: 6. N address=0x81091f8 frag=0x81091f0 size=16 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[645]: 71. N address=0x810a6f8 frag=0x810a6f0 size=128 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[645]: 73. N address=0x810a818 frag=0x810a810 size=128 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[645]: 74. N address=0x810a8a8 frag=0x810a8a0 size=128 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[645]: 75. N address=0x810a938 frag=0x810a930 size=128 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[645]: 76. N address=0x810a9c8 frag=0x810a9c0 size=128 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[645]: 77. N address=0x810aa58 frag=0x810aa50 size=128 used=1 Jun 26 14:37:56 globalser /usr/local/sbin/ser[645]: 78. N address=0x810aae8 frag=0x810aae0 size=128 used=1

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Re: [Users] Cisco radius auth
by Glenn Dalgliesh 30 Jun '06

30 Jun '06

What is your existing billing system? Many radius based platforums support digest auth. The cisco portion take some coding to make work but can be done. --------------------- Sent with ChatterEmail True push email for the Treo Smartphone www.chatteremail.com -----Original Message----- From: Jeff Williams <jeffw(a)globaldial.com> Date: Thursday, Jun 29, 2006 9:55 pm Subject: Re: [Users] Cisco radius auth Jeff Williams wrote: Is there any way to get openser to do radius auth using attribute from the Cisco VSA Voice Implementation Guide <http://www.cisco.com/en/US/products/sw/iosswrel/ps1835/products_programming…>? i.e. something like: > Access-Request NAS-IP-Address = 203.56.98.120 NAS-Port-Type = Async User-Name = "62186234" cisco-h323-conf-id = "h323-conf-id=89437006 E20011DA B677BBCA D2EBCDD5" Called-Station-Id = "0011553334112401" Calling-Station-Id = "0295731743" User-Password = "<145>;<22><242>$<251><243>)u<183>h<225>y<173><150><193>" > I would like to get openser to authenticate against an existing billing system if possible. Mmm, I realised after sending this that since SIP does digest auth and doesn't send a password as such, this is probably not possible. Jeff _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

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[Serusers] Assure a BYE is received ??
by Adrian Georgescu 30 Jun '06

30 Jun '06

There is no guarantee that a BYE will be received from an end-point. One possible solution is to use MediaProxy 1.7.2 in combination with SER. It will correctly terminate calls that have no BYE, by updating the radacct table AcctSessionTime column based on the last time RTP was relayed. Adrian ===== Hi, Use for logging: if (method=="BYE" || method=="CANCEL") { log(1, "SER: BYE"); setflag(1); } and record_route() to be sure that "BYE" wil visit your server. Dani On Mon, 26 Jun 2006 17:42:24 +0800 Ryan Pagquil <rpagquil at philonline.com> wrote: > Hi, > How can I be sure that all calls are terminated by a BYE message? I > have some instances that a BYE message is not logged by SER. Is there > a way to fix this? What could be the possible scenarios that causes > missing BYE's? > > Thanks, > Ryan >

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[Users] Using # for Sip 2 Sip calls
by Glenn Dalgliesh 29 Jun '06

29 Jun '06

Give this a try.. also if issues still exist include log messages from a call attempt with the ngrep. If you get a 407 and your client is getting it back you should see a second invite including the auth info. debug=3 fork=yes log_stderror=no listen=192.168.1.6 alias=test.com port=5060 children=4 dns=no rev_dns=no fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") 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") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); exit; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); t_relay(); exit; } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { route(1); return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); return; }; if (method=="ACK") { route(1); return; } else if (method=="CANCEL") { route(1); return; } else if (method=="INVITE") { xlog("Invite section\n"); route(3); return; } else if (method=="REGISTER") { route(2); return; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; }; route(3); route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); exit; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); return; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- xlog("Route 3\n"); if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); } if (uri=~"^sip:\*[0-9]*@"){ xlog("Sip 2 Sip\n"); strip(1); #strip away the * lookup(aliases); lookup(locations); route(4); route(1); return; } if (!lookup("location")){ if (uri=~"^sip:[0-9]*@") { xlog("PSTN Gateway\n"); route(4); route(5); return; }; sl_send_reply("404", "User Not Found"); return; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); } } route[5] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- xlog("Routed to route 5\n"); rewritehostport("pstn.gateway:5060"); avp_write("i:45", "inv_timeout"); route(1); } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; }

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Re: [Users] Using # for Sip 2 Sip calls
by Kenny Chua 29 Jun '06

29 Jun '06

So I took your advice and decided to use * to identify sip 2 sip calls. However, theres something wrong with my routing. I added route(6) to get authorize. Because when I try to dial sip to sip I get 407 proxy authentication required. Still after adding route(6), I still get the 407 proxy authentication required message. What is wrong? Route (1) is just the default message handler This is what I have: route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (uri=~"^sip:\*[0-9]*@"){ xlog("Sip 2 Sip\n"); strip(1); #strip the * because we dont need it route(4); route(6); route(1); return; }; if (!lookup("location")){ if (uri=~"^sip:[0-9]*@") { # International PSTN xlog("PSTN Gateway\n"); route(4); route(5); return; }; sl_send_reply("404", "User Not Found"); return; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); } } route[5] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- xlog("Routed to route 5\n"); rewritehostport("pstn.gateway:5060"); avp_write("i:45", "inv_timeout"); route(1); } route[6] { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: Hi, that's right. For example SIPURA ATAs with two lines but online one terminal use # for line selection.... you better use a digit that does not overlap with the PSTN dialling plan. regards, bogdan Glenn Dalgliesh wrote: >Well I would becarefull using # since some UA's use # to terminate digit input and dial..... Not positive but I think * would be a better choice. >--------------------- >Sent with ChatterEmail >True push email for the Treo Smartphone >www.chatteremail.com > > >-----Original Message----- >From: Kenny Chua >Date: Wednesday, Jun 28, 2006 10:56 pm >Subject: [Users] Using # for Sip 2 Sip calls > >Hello, I was wondering how to set my dialing plans to use # only for Sip 2 Sip calls. A user has to press the # sign if he wants to call another sip number, and just dial normally for PSTN calls? > > I came up with something like this: > lookup("aliases"); > if (uri=~"^sip:#[0-9]*@"){ > xlog("Sip 2 SIP\n"); > route(4); > route(1); > return; > }; > > Which of course don't work. So I'll need help. I know its possible to use 9 for PSTN calls, but I'm sure that you can use # for Sip 2 Sip. Please help me out here. Thank you. > > >--------------------------------- >Do you Yahoo!? > Get on board. You're invited to try the new Yahoo! Mail Beta. >--0-591390942-1151549737=:48905 >Content-Type: text/html; charset=iso-8859-1 >Content-Transfer-Encoding: 8bit > >Hello, I was wondering how to set my dialing plans to use # only for Sip 2 Sip calls. A user has to press the # sign if he wants to call another sip number, and just dial normally for PSTN calls? I came up with something like this: lookup("aliases"); if (uri=~"^sip:#[0-9]*@"){ xlog("Sip 2 SIP\n"); route(4); route(1); return; }; Which of course don't work. So I'll need help. I know its possible to use 9 for PSTN calls, but I'm sure that you can use # for Sip 2 Sip. Please help me out here. Thank you. > --------------------------------- Sneak preview the all-new Yahoo.com. It's not radically different. Just radically better.

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[Serusers] sip phone for solaris?
by Cesc 29 Jun '06

29 Jun '06

Hi, Any experiences? ideally it would come pre-compiled, but anything will be good enough (own compilation ... Java ... ) Cesc

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[Users] avp problems switching from 1.0.1 to 1.1.x
by tele 29 Jun '06

29 Jun '06

Hi, My billing system doesn't correlate START/STOP accounting packets and use Cisco-VSA like attributes. I need to do an accounting compatible with my billing system from OpenSER so i've done a dirty method like that: i know it's not really performance but it's the only way i can do that. The problem is with OpenSER 1.1.x that generate accounting with the attribute filled with the same value of "disconnect_time". example: h323-setup-time = '1151511700' h323-connect-time = '1151511700' h323-disconnect-time = '1151511700' the xlog debug output of openser1.1.x i have added is with correct attribute values. with OpenSER 1.0.1 everything works fine. CONFIGS: with OpenSER 1.1.x: modparam("acc", "radius_extra", "h323-setup-time=$avp(setup_time); h323-connect-time=$avp(connect_time); h323-disconnect-time=$avp(disconnect_time)") modparam("avpops","avp_aliases","setup_time=s:setup_time") modparam("avpops","avp_aliases","connect_time=s:connect_time") modparam("avpops","avp_aliases","disconnect_time=s:disconnect_time") if (is_method("INVITE")) { avp_write("$Ts","$avp(setup_time)"); avp_db_store("$from","$avp(setup_time)"); } else if (is_method("ACK")) { avp_write("$Ts","$avp(connect_time)"); avp_db_store("$from","$avp(connect_time)"); } else if (is_method("BYE")) { avp_write("$Ts","$avp(disconnect_time)"); avp_db_store("$from","$avp(disconnect_time)"); }; if (is_method("BYE")) { avp_db_load("$from","$avp(setup_time)"); avp_db_load("$from","$avp(connect_time)"); avp_db_load("$from","$avp(disconnect_time)"); xlog("L_INFO","Before accounting SETUP_TIME[$avp(setup_time)] CONNECT_TIME[$avp(connect_time)] DISCONNECT_TIME[$avp(disconnect_time)]"); setflag(3); setflag(5); avp_db_delete("$from","$avp(setup_time)"); avp_db_delete("$from","$avp(connect_time)"); avp_db_delete("$from","$avp(disconnect_time)"); }; With OpenSER 1.0.1: modparam("acc", "radius_extra", "h323-setup-time=$avp($setup_time); h323-connect-time=$avp($connect_time) ; h323-disconnect-time=$avp($disconnect_time)") modparam("avpops","avp_aliases","setup_time=s:setup_time") modparam("avpops","avp_aliases","connect_time=s:connect_time") modparam("avpops","avp_aliases","disconnect_time=s:disconnect_time") if (is_method("INVITE")) { avp_write("0","i:1"); avp_printf("i:1", "$Ts"); avp_copy("i:1","$setup_time"); avp_db_store("$from","$setup_time"); avp_delete("i:1"); } else if (is_method("ACK")) { avp_write("0","i:1"); avp_printf("i:1", "$Ts"); avp_copy("i:1","$connect_time"); avp_db_store("$from","$connect_time"); avp_delete("i:1"); } else if (is_method("BYE")) { avp_write("0","i:1"); avp_printf("i:1", "$Ts"); avp_copy("i:1","$disconnect_time"); avp_db_store("$from","$disconnect_time"); avp_delete("i:1"); }; if (is_method("BYE")) { avp_db_load("$from","$setup_time"); avp_db_load("$from","$connect_time"); avp_db_load("$from","$disconnect_time"); xlog("L_INFO","Before accounting SETUP_TIME[$avp($setup_time)] CONNECT_TIME[$avp($connect_time)] DISCONNECT_TIME[$avp($disconnect_time)]"); setflag(3); setflag(5); avp_db_delete("$from","$setup_time"); avp_db_delete("$from","$connect_time"); avp_db_delete("$from","$disconnect_time"); };

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