- sr-users - kamailio.org

[Users] loose_route() (misbehaviour?)
by samuel 25 Jan '06

25 Jan '06

Hi all, Might be related to the mail I just sent about the username in the record-route or to another problem but the thing is that I can not close the session from the callee side...i'll try to explain: There are two users registered to two different openser instances. In the session initiation, the record route headers contain the usernames of both users and are therefore stored in the end points. When the caller sends the BYE, the callee's openser receives the BYE and rewrites the URI with the record-route (shouldn't it consider the request for itself and check the Req-URI instead??). NExt is the diagram and some more info... Alice oser1 oser2 Bob | | | | |--F1---> |--F2--->|--F3-->| | | | | | | | | | | | | |--F4--> |--F5-->|--F6--> | | | | | | | | | | | | | | |<--F8---|<--F9--| F1: INVITE Bob@oser2 F2: INVITE Bob@oser2 Record-Route:<sip:Bob@oser1;nat=yes;lr=on> F3: INVITE Bob@bob_contact_ip Record-Route:<sip:Bob@oser2;nat=yes;lr=on> Record-Route:<sip:Bob@oser1;nat=yes;lr=on> OK,ACK not shown.. F4: BYE Bob@nat_fixed_ip Route:<sip:Bob@oser1;nat=yes;lr=on>,<sip:Bob@oser2;nat=yes;lr=on> F5: BYE Bob@nat_fixed_ip Route:<sip:Bob@oser1;nat=yes;lr=on> F6: BYE Bob@bob_fixed_ip Everything ok, session closed. But, and here comes the problem: F7: BYE Alice@alice_fixed_ip Route:<sip:Bob@oser2;nat=yes;lr=on>,<sip:Bob@oser1;nat=yes;lr=on> When this message arrives to ser1, it sees the route header and, make a strict routing process and rewrites the header in the Req-URI to Bob@ser1. oser1 resends itself the message, and when it receives it again looks for use Bob in the local bindings which obviously does not exists (Bob is in oser2). Log: Jan 25 13:17:07 localhost openser[25501]: is_preloaded: No Jan 25 13:17:07 localhost openser[25501]: grep_sock_info - checking if host==us: 13==13 && [ip_oser1]== [ip_oser1] Jan 25 13:17:07 localhost openser[25501]: grep_sock_info - checking if port 5060 matches port 5060 Jan 25 13:17:07 localhost openser[25501]: after_strict: Next hop: 'sip:Bob@oser1_ip;nat=yes;lr=on' is loose router Jan 25 13:17:07 localhost openser[25501]: parse_headers: flags=ffffffffffffffff Jan 25 13:17:07 localhost openser[25501]: DEBUG: get_hdr_body : content_length=0 Jan 25 13:17:07 localhost openser[25501]: found end of header Jan 25 13:17:07 localhost openser[25501]: rewrite_uri: Rewriting Request-URI with 'sip:Bob@oser1_ip:5060;nat=yes;lr=o n' Jan 25 13:17:07 localhost openser[25501]: after_strict: The last route URI: 'sip:Bob@oser1_ip:5060;nat=yes;lr=on' Jan 25 13:17:07 localhost openser[25501]: DEBUG: has_totag: totag found Jan 25 13:17:07 localhost openser[25501]: DEBUG: t_newtran: msg id=12 , global msg id=11 , T on entrance=0xffffffff Jan 25 13:17:07 localhost openser[25501]: parse_headers: flags=ffffffffffffffff Jan 25 13:17:07 localhost openser[25501]: parse_headers: flags=78 Jan 25 13:17:07 localhost openser[25501]: t_lookup_request: start searching: hash=3891, isACK=0 Jan 25 13:17:07 localhost openser[25501]: DEBUG: RFC3261 transaction matching failed Jan 25 13:17:07 localhost openser[25501]: DEBUG: t_lookup_request: no transaction found Thanks a lot!!! Sam.

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[Users] Question concerning nathelper and rtpproxy
by Wolfgang Hottgenroth 25 Jan '06

25 Jan '06

Hi, I'm trying to get nathelper and rtpproxy to work, following the guide found on onsip.org, but unfortunately, it does not work correctly for. Don't know why. My setup is the following: SIP-Phone (170.127.120.189) -> Proxy1 (62.191.185.30) -> Proxy2 (62.191.185.147) -> PSTN-GW (139.4.200.6) Proxy1 supports authentication and handles the NAT travseral stuff. Since in the current testbed, SIP-Phone is not actual behind a NAT device I bypassed the test in the openser configuration to recognize this phone as "nated". What I see is, for a not nated phone, that the signaling goes the chain as drawn above and the media stream goes from the phone to the ngw and vice-versa, as I've expected. Everything is fine in this case. For a nated phone, the signaling again goes through the chain, but the media stream from the phone goes to the rtpproxy on Proxy1, which tries to send it to Proxy2, where no rtpproxy is running, and thereby an "ICMP port unreachable" is returned (which actual does prevent the rtpproxy from continously sending the media stream). Digging into the SIP and SDP is see the following: SIP-Phone sends an INVITE to Proxy1, in the SDP it has "Connection Information: 170.127.120.189" (IP of the phone) Proxy1 replies with "100 Trying" and sends the INVITE to Proxy2, in the SDP it has now "Connection Information: 62.191.185.30" (IP of Proxy1), as I've expected. Proxy2 replied with "100 Trying" and sends the INVITE to the ngw, in the SDP it still has "Connection Information: 62.191.185.30". Now the ngw sends "183 Session Progress" to Proxy2, which has in the SDP "Connection Information: 139.4.200.6" (IP of the ngw). Proxy2 sends "183 Session Progress" to Proxy1, still with "Connection Information: 139.4.200.6" in the SDP. Finally Proxy1 sends "183 Session Progress" to the phone, now with "Connection Information: 62.191.185.30" (IP of Proxy1) in the SDP, as I've expected, since the phone should send the media stream through the rtpproxy running on Proxy1. The phone sends the media stream as expected to Proxy1. But, and this is the problem, the rtpproxy sends the media stream not to the ngw, as I've expected looking into "SDP Connection Information" received by this box, but to Proxy2, which never was mentioned in a Connection Information. When starting rtpproxy with -f, I saw the it was asked by openser/nathelper to proxy between the phone and Proxy2, so the rtpproxy seems to work correctly. But why asks the nathelper on Proxy1 to proxy the media between the phone and Proxy2 and not, as I've expected between the phone and the ngw? My openser configuration is attached, if someone would be so nice to have a look and send a hint for me. Thanks, Wolfgang # # $Id: openser.cfg,v 1.5 2005/10/28 19:45:33 bogdan_iancu Exp $ # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (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/openser_fifo" # ------------------ module loading ---------------------------------- loadmodule "/opt/openser/lib/openser/modules/sl.so" loadmodule "/opt/openser/lib/openser/modules/tm.so" loadmodule "/opt/openser/lib/openser/modules/rr.so" loadmodule "/opt/openser/lib/openser/modules/maxfwd.so" loadmodule "/opt/openser/lib/openser/modules/usrloc.so" loadmodule "/opt/openser/lib/openser/modules/registrar.so" loadmodule "/opt/openser/lib/openser/modules/textops.so" loadmodule "/opt/openser/lib/openser/modules/auth.so" loadmodule "/opt/openser/lib/openser/modules/auth_radius.so" loadmodule "/opt/openser/lib/openser/modules/xlog.so" loadmodule "/opt/openser/lib/openser/modules/nathelper.so" loadmodule "/opt/openser/lib/openser/modules/uri.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) #modparam("rr", "enable_full_lr", 1) modparam("xlog", "buf_size", 8192) modparam("auth_radius", "radius_config", "/opt/openser/etc/openser/radiusclient.conf") modparam("auth", "rpid_prefix", "<sip:") modparam("auth", "rpid_suffix", "@62.191.185.30>;screen=yes;privacy=off") modparam("auth", "rpid_avp", "rpid") modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/opt/rtpproxy/sock/rtpproxy.sock") # ------------------------- request routing logic ------------------- # main routing logic route{ xlog("L_NOTICE", "REQUEST: $rm, $ru, $si\n"); if (method != "REGISTER") { record_route(); } if (method == "BYE" || method == "CANCEL") { unforce_rtp_proxy(); } if (src_ip == "170.127.120.189") { xlog("L_NOTICE", "*** FOUND THAT PHONE\n"); setflag(6); force_rport(); } if (loose_route()) { xlog("L_NOTICE", "*** loose routing enforced\n"); if (has_totag() && (method == "INVITE" || method == "ACK")) { if (nat_uac_test("3")) { xlog("L_NOTICE", "*** found NAT (1) !\n"); setflag(6); force_rport(); fix_nated_contact(); } force_rtp_proxy("l"); } route(1); return; } if (uri != myself) { xlog("L_NOTICE", "*** uri != myself\n"); sl_send_reply("483", "will not handle this"); return; } if (method == "CANCEL") { xlog("L_NOTICE", "*** cancel\n"); route(1); return; } else if (method == "INVITE") { xlog("L_NOTICE", "*** invite\n"); route(3); return; } else if (method == "REGISTER") { xlog("L_NOTICE", "*** register\n"); route(2); return; } xlog("L_NOTICE", "*** at the end: $rm\n"); route(1); return; } route[1] { t_on_reply("1"); if (!t_relay()) { if (method == "INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); } } route[2] { if (!search("^Contact: [ ]*\*") && nat_uac_test("3")) { xlog("L_NOTICE", "*** found NAT (2) !\n"); if (nat_uac_test("1")) { xlog("L_NOTICE", " --> 1\n"); } if (nat_uac_test("2")) { xlog("L_NOTICE", " --> 2\n"); } setflag(6); fix_nated_register(); force_rport(); } if (! radius_www_authorize("uu.net")) { www_challenge("uu.net", "0"); return; } consume_credentials(); if (!save("location")) { sl_reply_error(); } return; } route[3] { if (src_ip == "62.191.185.147") { xlog("L_NOTICE", "*** from upstream proxy\n"); if (uri =~ "^sip:99") { xlog("L_NOTICE", "*** outbreak prefix from outside, reject\n"); sl_send_reply("484", "no pstn outbreak for you"); return; } } else { if (! radius_www_authorize("uu.net")) { www_challenge("uu.net", "0"); return; }; consume_credentials(); if (nat_uac_test("3")) { xlog("L_NOTICE", "*** found NAT (3) !\n"); setflag(6); } append_rpid_hf(); if (uri=~"^sip:99") { xlog("L_NOTICE", "*** pstn outbreak call, uri: $ru\n"); strip(2); xlog("L_NOTICE", "*** stripped uri: $ru\n"); if (uri=~"^sip:[1-9]") { xlog("L_NOTICE", "*** local call\n"); prefix("+49231"); } else if (uri=~"^sip:0[1-9]") { xlog("L_NOTICE", "*** national call\n"); strip(1); prefix("+49"); } else if (uri=~"^sip:00[1-9]") { xlog("L_NOTICE", "*** international call\n"); strip(2); prefix("+"); } else { xlog("L_NOTICE", "*** something wrong\n"); sl_send_reply("488", "wrong telephone number"); return; } rewritehostport("62.191.185.147:5060"); route(4); route(1); return; } } xlog("L_NOTICE", "*** looking up ...\n"); if (!lookup("location")) { xlog("L_NOTICE", " --> not found\n"); sl_send_reply("404", "Not Found"); return; } xlog("L_NOTICE", " --> found\n"); route(4); route(1); return; } route[4] { if (isflagset(6)) { xlog("L_NOTICE", "*** handle NAT (1)\n"); force_rport(); fix_nated_contact(); force_rtp_proxy(); } return; } onreply_route[1] { xlog("L_NOTICE", "*** onreply_route\n"); if (isflagset(6) && status =~ "(180)|(183)|2[0-9][0-9]") { if (! search("^Content-Length:[ ]*0")) { xlog("L_NOTICE", "*** handle NAT (2)\n"); force_rtp_proxy(); } } if (nat_uac_test("1")) { fix_nated_contact(); xlog("L_NOTICE", "*** handle NAT (3)\n"); } return; }

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[Users] Re: [Devel] web interface development
by Bogdan-Andrei Iancu 25 Jan '06

25 Jan '06

I put also "users" at CC - more feedback is welcomed ;). wiki location: http://openser.org/dokuwiki/doku.php?id=openser_web_interface_documentation regards, bodgan Mike Williams wrote: >I've started posting a few design for the web interface plans on the Doku Wiki, they are in an early stage and I will continue to add to them. Please discuss or ask questions on the development list. Also, I would like to see people start adding wanted features and goals to a Wiki Page. > >Thanks. > >---Mike > >_______________________________________________ >Devel mailing list >Devel(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/devel > > >

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[Users] enum patch
by Greg Fausak 25 Jan '06

25 Jan '06

Openser users, ------ is_from_user_enum: I find this extremely useful. This function uses the 'From' uri and does a enum lookup based on that number. A true or false is returned. I route to my customers using enum, so when the customer does an outbound call i can use this to verify that they are indeed a valid customer before routing the call. enum_fquery: Carry the previous function another step. You can do 'source routing' of a call using the from address. I do a enum lookup based on the from number, and then apply the ruri to number to the naptr that was found. Using this technique you can 'source route' individual numbers. In this way, for example, customer A can elect (pay for) routing out 2 or 3 gateways, while customer B elects to just have one provider. Another interesting feature is to temporarily reroute a customer to an intercept stating that his account is suspended. -------- Both of these functions assume that the openser admin is operating private enum domain databases. I haven't created a patch file before. This file is a context diff output for a few files in the modules/enum directory. -g

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Re: [Users] Crash (core)
by Bogdan-Andrei Iancu 25 Jan '06

25 Jan '06

Hi Lars, it took some time, but thanks to your details debug output I was able to trace the problem and fix the bug. As I was expecting, in was in the to/from parser. Please update from CVS and give it a try. Thanks again for the help, bogdan Lars Sundqvist wrote: > I wish I could, but the INVITE's are not logged and I have not been > able to reproduce the problem on the test system :-( so i realize this > isn't very easy to "fix" I was mainly checking if this was a known > problem. > > Version is 1.0.0-tls (disable_tls=0) compiled from source. > > uac is called like: > uac_replace_from("SomeText: $avp(s:display)","") > > The INVITE was most likely produced by a "Aastra Intelligate" SIP UA > since the particular customer has one of these registered right now. > > /Lasse > > Bogdan-Andrei Iancu wrote: > >> Hi Lars, >> >> can you please post the complete INVITE request that produces the >> crash and how the uac_replace_from() function is called from script? >> and what version are you using? >> >> regards, >> bogdan >> >> Lars Sundqvist wrote: >> >>> I have switched from SER to OpenSER in our production enviroment >>> since I >>> need a few of the "new" modules, I am however having troubles with >>> OpenSER hanging and crashing, so much so, that I had to change back. >>> >>> There is no clues in the logs to why this is happening but i have a >>> coredump from a crash: >>> >>> (gdb) where >>> #0 0x0052f7a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 >>> #1 0x005707d5 in raise () from /lib/tls/libc.so.6 >>> #2 0x00572149 in abort () from /lib/tls/libc.so.6 >>> #3 0x08054863 in del_lump (msg=0x0, offset=186, len=-135181242, >>> type=HDR_OTHER_T) at data_lump.c:288 >>> #4 0x00f8bd20 in replace_from (msg=0x8130d98, from_dsp=0xbff977a0, >>> from_uri=0x0) at from.c:207 >>> #5 0x00f8e57b in w_replace_from2 (msg=0x8130d98, dsp=0x812f700 >>> "\030W\022\b\v", uri=0x0) at uac.c:311 >>> #6 0x08051703 in do_action (a=0x8126658, msg=0x8130d98) at >>> action.c:677 >>> #7 0x080529e3 in do_action (a=0x8126680, msg=0x8130d98) at >>> action.c:773 >>> #8 0x0805297b in do_action (a=0x8126a58, msg=0x8130d98) at >>> action.c:773 >>> #9 0x0805297b in do_action (a=0x8126f80, msg=0x8130d98) at >>> action.c:773 >>> #10 0x0805297b in do_action (a=0x8127058, msg=0x8130d98) at >>> action.c:773 >>> #11 0x0805297b in do_action (a=0x812b6d0, msg=0x8130d98) at >>> action.c:773 >>> #12 0x080529e3 in do_action (a=0x812b6f8, msg=0x8130d98) at >>> action.c:773 >>> #13 0x080529e3 in do_action (a=0x812b720, msg=0x8130d98) at >>> action.c:773 >>> #14 0x0805297b in do_action (a=0x812b7f8, msg=0x8130d98) at >>> action.c:773 >>> #15 0x08052b42 in run_actions (a=0x8122da0, msg=0x8130d98) at >>> action.c:773 >>> #16 0x080753e9 in receive_msg ( >>> buf=0x80eb300 "INVITE sip:987654321@xxx.xxx.xxx SIP/2.0\r\nVia: >>> SIP/2.0/UDP >>> 192.168.0.17:5060;rport;branch=z9hG4bK0ffd9a6274aeb347617736a6da35e918\r\nTo: >>> >>> 987654321<sip:987654321@xxx.xxx.xxx>\r\nFrom: 123456789<si"..., >>> len=943, >>> rcv_info=0xbff98b80) >>> at receive.c:160 >>> #17 0x0808dd1c in udp_rcv_loop () at udp_server.c:464 >>> #18 0x08064e7f in main_loop () at main.c:1033 >>> #19 0x08065615 in main (argc=3, argv=0xbff98d34) at main.c:1570 >>> >>> (Numbers and sip domain edited) >>> Anyone who has a clue whats going wrong? >>> >>> Lars Sundqvist >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)openser.org >>> http://openser.org/cgi-bin/mailman/listinfo/users >>> >> >

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SV: [Serusers] Setting up ser (configuring static forwarding)
by Robert Zettinger 25 Jan '06

25 Jan '06

Hi, I am running everything on machines on the same private network. My softphone is on a windows machine with the local ip address 192.168.1.183, ser is on a Linux machine with the local ip address 192.168.1.147, and my friends softphone is on a windows machine with the local ip address 192.168.1.192. And what I am trying to do is: 1. Having my softphone use 192.168.1.147 (the ser machine) as proxy. 2. Setting ser to forward a call to cooldude(a)company.se to 192.168.1.192 (my friend with a softphone) Robert -----Ursprungligt meddelande----- Från: Ladislav Andel [mailto:ladia6@centrum.cz] Skickat: den 25 januari 2006 13:51 Till: Robert Zettinger; serusers(a)lists.iptel.org Ämne: Re: [Serusers] Setting up ser (configuring static forwarding) Hi, you should better tell us where is your SER running . Inside of private network? public internet? Is your softphone in the same private network as your friend's softphone? Could you draw your network endpoints eg. SIP UA---(public internet)---- SER --- (private network) --- your friend's SIP UA If we know the location of your network entities then it would help us a lot to understand what you are trying to do. Ladislav Robert Zettinger wrote: > > I have problems setting up Sip Express Router (SER). I have installed > it, and it seems to run fine, and my softphone seems to able to > connect to it. Just to see if it works, I want to set up ser to > connect my softphone to my friend's (his IP is 192.168.1.192). > Therefore I change the ser.cfg file to look like this: > > ---------- > > # > # $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 "../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) > > # 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=="sip:cooldude@company.se") { > forward( 192.168.1.192, 5060); > } > > 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(); > }; > } > > > ---------- > > Where > > if (uri=="sip:cooldude@company.se") { > forward( 192.168.1.192, 5060); > } > > is the only thing I have added to the original default config file > that comes with the distribution. I now want to enter > "sip:cooldude@company.se" in my softphone and get forwarded to > 192.168.1.192. But nothing happens. Ideas? > > > > I am using Linux 2.4.18-14 and ser 0.9.6. > > ------------------------------------------------------------------------ > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

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[Users] rr add_username
by samuel 25 Jan '06

25 Jan '06

Hi all!!!! My openSER does not honour the add_username parameter of the rr module and always insert the user part in the record route headers. Maybe it is already corrected in the CVS... I downloaded the next snapshot openser-1.0.0-cvs-13Jan.tgz and the output from the -V is version: openser 1.0.0 (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, 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 1.7 2005/10/28 14:55:23 klaus_darilion Exp $ main.c compiled on 16:28:45 Jan 16 2006 with gcc 3.3 Thanks a lot, Sam.

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[Serusers] Setting up ser (configuring static forwarding)
by Robert Zettinger 25 Jan '06

25 Jan '06

I have problems setting up Sip Express Router (SER). I have installed it, and it seems to run fine, and my softphone seems to able to connect to it. Just to see if it works, I want to set up ser to connect my softphone to my friend's (his IP is 192.168.1.192). Therefore I change the ser.cfg file to look like this: ---------- # # $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 "../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) # 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=="sip:cooldude@company.se") { forward( 192.168.1.192, 5060); } 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(); }; } ---------- Where if (uri=="sip:cooldude@company.se") { forward( 192.168.1.192, 5060); } is the only thing I have added to the original default config file that comes with the distribution. I now want to enter "sip:cooldude@company.se" in my softphone and get forwarded to 192.168.1.192. But nothing happens. Ideas? I am using Linux 2.4.18-14 and ser 0.9.6.

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[Users] uac_replace_from Core
by H Quintana 25 Jan '06

25 Jan '06

Hi, I'd like to have some ideas for the solution of this problem: In OpenSer 1.0., when I use the function uac_replace_from("","me(a)here.com") I'm getting the following error: Jan 24 15:19:59 server1 /usr/local/sbin/openser[22100]: BUG: del_lump: offset + len exceeds message size (79 + -135152687 > 946) Jan 24 15:20:00 server1 /usr/local/sbin/openser[22084]: child process 22100 exited by a signal 6 Jan 24 15:20:00 server1 /usr/local/sbin/openser[22084]: core was generated Jan 24 15:20:00 server1 /usr/local/sbin/openser[22084]: INFO: terminating due to SIGCHLD Thanks for your help. Best regards, Humberto __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] rtpproxy
by Olivier Taylor 25 Jan '06

25 Jan '06

Hi all, I have a ser(0.9.4/Freebsd) server well working, but in some cases, I have a strang behavior. I use rtpproxy for nat traversal,all my users can call pstn, but when we try to call one of our users from internal or from pstn, Ser sends many invites and callee never answer to these invite. I must conclude that they don't receive it... All the users having that problems are under a nated network. Any idea ? Olivier

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[Serusers] Billing - Call Forward
by Bjorn Asmul 25 Jan '06

25 Jan '06

Has anyone successfully been able to set the "From"-field to the original "To"-field after changing the r-uri to the forwarded destination? A calls B which forwards to C Currently A is billed because that's the callerID that goes through. I want to bill B. -- Bjorn

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[Serusers] Radius AVOPS
by Deepak Singhal 25 Jan '06

25 Jan '06

Can the caller party IP Address be added in radius authentication and accounting packets for SER ? Perhaps using avops module ? Regards Deepak Singhal

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RE: [Serusers] Billing - Call Forward
by Bjorn Asmul 25 Jan '06

25 Jan '06

Don't think so. append_rpid_hf() maybe, but I haven't had any luck with it... -----Original Message----- From: Paul Hazlett [mailto:javarockx@gmail.com] Sent: Tuesday, January 24, 2006 11:28 PM To: Bjorn Asmul Cc: seruser Subject: Re: [Serusers] Billing - Call Forward I'm not too familiar with the UAC module, but perhaps it will do the trick. Cheers, Paul On 1/24/06, Bjorn Asmul <bjorn(a)atlasvoice.com> wrote: > > Has anyone successfully been able to set the "From"-field to the > original "To"-field after changing the r-uri to the forwarded destination? > > A calls B which forwards to C > > Currently A is billed because that's the callerID that goes through. > I want to bill B. > > -- Bjorn > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > >

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RE: [Serusers] re : presence module with ser
by Yeung OnTai-q16645 25 Jan '06

25 Jan '06

Hi Jan, I'd like to use PA, XCAP with eyeBeam. Are you talking about the "ser-0.10.99-dev30-tm-timers-pa-3", I've downloaded the tar file and tried "make install". It seems like presence works fine, XCAP is also working(except RLS)... About the RLS, do I need to create a directory in "<xcap-root>/rls-services/global/index" where all the XML (for RLS) files are stored? How can I configure the eyeBeam accordingly? By default it's configured as "subscribe to contact list : $username$-list@$domain$". It seems like SER does not consider it as a list, it only considers it just another URI. Did I do something wrong? Or do I need to change anything in SER? Regards, Dennis -----Original Message----- From: Jan Janak [mailto:jan@iptel.org] Sent: Tuesday, January 24, 2006 11:11 PM To: Yeung OnTai-q16645 Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] re : presence module with ser On 18-01-2006 09:07, Yeung OnTai-q16645 wrote: > Hi all, > > I need to build SER with presence and XCAP. Does anyone know which > version of SER contain a relatively "stable" version of "pa" and "rls". > I've tried 0.9.6 and the cvs stable build, but I ran into different > problems(error in loading modules, compling errors, etc) in making the > source code. > > Can anyone point me to the right direction please? Try the version from http://ftp.iptel.org/pub/ser/presence That one should be relatively easy to build. Note that the XCAP server is not included. Jan. > > Thank you very much! > > Regards, > Dennis > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

1 0

Re: Fwd: [Serusers] using dbtext module for authentication
by Jan Janak 25 Jan '06

25 Jan '06

It seems that you do not have "version" table. that table contains version of each table in your database, this is used for sanity checking. You can take a look into ser_mysql.sh and copy the contents of version table from corresponding sql table. Jan. On 20-01-2006 16:08, Yi Zheng wrote: > Hi, > > I have the following authentication related stuff in my ser.cfg file. However ser server failed to start with the following messages logged. I have a directory /root/dbtext/ser to store tables (such as subscriber). What could be wrong? Thanks, > > log messages > -------------------- > Jan 20 13:44:08 my ./ser[4424]: table_version(): Error in db_query > Jan 20 13:44:08 my ./ser[4424]: authdb_fixup: Error while querying table > version > Jan 20 13:44:08 my ./ser[4424]: ERROR: fix_expr : fix_actions error > > > > My ser.cfg > --------------- > loadmodule "/home/ser/presence/ser-0.10.99-dev30-tm-timers-pa-3/lib/ser/modules/dbtext.so" > loadmodule "/home/ser/presence/ser-0.10.99-dev30-tm-timers-pa-3/lib/ser/modules/auth.so" > loadmodule "/home/ser/presence/ser-0.10.99-dev30-tm-timers-pa-3/lib/ser/modules/auth_db.so" > > # -- 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) > modparam("usrloc", "db_mode", 0) > modparam("pa", "use_db", 0) > modparam("auth_db", "db_url", "dbtext:///root/dbtext/ser") > > /root/dbtext/ser/subscriber > ------------------------------------- > username(str) password(str) ha1(str) domain(str) ha1b(str) > suser:supasswd:xxx:iptel.org:xxx > test:test123::my.cisco.com:: > > Note: forwarded message attached. > Date: Fri, 20 Jan 2006 11:36:18 -0800 (PST) > From: Yi Zheng <yizheng(a)sbcglobal.net> > Subject: [Serusers] using dbtext module for authentication > To: serusers(a)lists.iptel.org > > Hi, > > I don't have enough space for MySql and hence want to try to use a plain text file (dbtext module) for authentication purpose. Just wondering whether it works properly with SER and what are the limitations besides limited space. Thanks, > > - ming > _______________________________________________ > 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

2 1

Re: [Serusers] re : add subscriber issue in ser-0.10.99
by Jan Janak 24 Jan '06

24 Jan '06

On 19-01-2006 17:43, Yeung OnTai-q16645 wrote: > When I use "#serctl add <username> <password> <email>", it does not work > anymore, because it complains that subscriber table is not found. serctl (the shell script) will be deprecated. > Whatever database table(subscriber or other) is being used now, there > should be a place where I can provision a new client with password. > Otherwise how can I authenticate it when the client register to SER? The table is called credentials, there is a command called ser_cred in sip_router/tools/serctl which can be used to provision the table. Note that an interface similar to the original serctl is being worked on, it will be available soon. > It seems like in ser-0.10.99, clients can just register to ser without > any provisioning.... No, but the newer SER version has much cleaner data model and the meaning of tables changed. credentials table maps digest credentials to users (identified by uid). uri table maps SIP URIs to users, and so on. Jan.

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[Serusers] avp_radius - how to start?
by Zoran Milic 24 Jan '06

24 Jan '06

Hello, Sorry for asking such a dumb question but I'm in a kind of a hurry. How do I declare variables in which I wish to store data I get from avp_load_radius() ? How do I assign a value to a variable and then check it up to some other value? Basicly my problem comes to avp+radius basic usage. Best regards, Zoran

1 0

[Serusers] NAT travesal: force_rport() what does it do?
by Yi Zheng 24 Jan '06

24 Jan '06

Hi, I am reading to understand the different solutions for NAT travesal. For nathelper+RTP proxy solution, what does the force_rport() function do? Is it rfc3581 compliant? Does it mean that the client needs to understand "received" and "rport" in the Via header? Thanks, - ming

2 1

[Serusers] SER version
by Voipers Portugal 24 Jan '06

24 Jan '06

I would like to know what's the most stable version of SER to use RADIUS, POSTGRES, LDAP, ENUM, LRC, NATHELPER and ACC modules? And what´s the name that i should use to get it from the CVS server. Regards, Jose Simoes

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RE: [Serusers] msilo, voicemail and serweb
by Alejandro Mellado 24 Jan '06

24 Jan '06

Yes, On serweb acount tab message store. I've seen the voicemail plugin code, and this doesn't write on mysql silo table. How I do put voicemail on serweb? Regards Alejandro. El sáb, 23-07-2005 a las 11:46 +0200, harry gaillac escribió: > Hello, > > You mean forward to voicemail on account tab ? > > harry > --- Alejandro Mellado <amellado(a)uct.cl> a écrit : > > > Hi > > > > > > I'm trying to activate the voicemail in serweb. > > I've working msilo with IM successfully. > > > > Somebody Can help me? > > > > > > Regards > > Saluda Atte. > > Alejandro Mellado <amellado(a)uct.cl>. > > Escuela de Informática > > Universidad Católica de Temuco > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > ___________________________________________________________________________ > Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger > Téléchargez cette version sur http://fr.messenger.yahoo.com Saluda Atte. Alejandro Mellado <amellado(a)uct.cl>. Escuela de Informática Universidad Católica de Temuco

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Re: [Serusers] using dbtext module for authentication
by Jan Janak 24 Jan '06

24 Jan '06

On 20-01-2006 11:36, Yi Zheng wrote: > Hi, > > I don't have enough space for MySql and hence want to try to use a plain text file (dbtext module) for authentication purpose. Just wondering whether it works properly with SER and what are the limitations besides limited space. Thanks, Not all modules have been tested with dbtext, you will probably have to try it for yourself. dbtext is slow, thus it is not suitable for larger setups. Updates of dbtext files are not realtime, thus you would not see changes immediately. Jan.

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Re: [Serusers] re : presence module with ser
by Jan Janak 24 Jan '06

24 Jan '06

On 18-01-2006 09:07, Yeung OnTai-q16645 wrote: > Hi all, > > I need to build SER with presence and XCAP. Does anyone know which > version of SER contain a relatively "stable" version of "pa" and "rls". > I've tried 0.9.6 and the cvs stable build, but I ran into different > problems(error in loading modules, compling errors, etc) in making the > source code. > > Can anyone point me to the right direction please? Try the version from http://ftp.iptel.org/pub/ser/presence That one should be relatively easy to build. Note that the XCAP server is not included. Jan. > > Thank you very much! > > Regards, > Dennis > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

1 0

Re: [Serusers] I have a problem: undefined symbol: dbf
by Jan Janak 24 Jan '06

24 Jan '06

It looks like you are trying to call the functions of DB API but do not have db_func_t defined anywhere in the module. Check, for example, the sources of auth_db module, look for db_func_t Jan. On 17-01-2006 14:47, jvr_78(a)yahoo.com.ar wrote: > Hi, > I'm new in this group. > I have work with ser 0.8.14 and now I need presence module, so actualice the > ser version to 0.10.99, and now don't work my module. > > the error that show is: > > #ser > 0(3981) ERROR: load_module: could not open module > </usr/local/lib/ser/modules/control.so>: /usr/local/lib/ser/modules/control.so: > undefined symbol: dbf > 0(3981) parse error (47,13-14): failed to load module > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (62,20-21): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (63,20-21): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (66,21-22): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (67,19-20): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (68,19-20): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (69,20-21): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (70,20-21): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (71,21-22): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (72,20-21): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (73,20-21): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (74,20-21): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (75,20-21): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (76,20-21): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (77,20-21): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (78,19-20): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (79,19-20): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (80,20-21): Can't set module parameter > 0(3981) parse error (130,28-29): unknown command, missing loadmodule? > > 0(3981) parse error (134,20-21): unknown command, missing loadmodule? > > 0(3981) parse error (150,33-34): unknown command, missing loadmodule? > > 0(3981) parse error (159,35-36): unknown command, missing loadmodule? > > 0(3981) parse error (161,25-26): unknown command, missing loadmodule? > > 0(3981) parse error (169,24-25): unknown command, missing loadmodule? > > 0(3981) parse error (184,42-43): unknown command, missing loadmodule? > > 0(3981) parse error (215,26-27): unknown command, missing loadmodule? > > 0(3981) parse error (219,24-25): unknown command, missing loadmodule? > > 0(3981) parse error (230,24-25): unknown command, missing loadmodule? > > 0(3981) parse error (263,28-29): unknown command, missing loadmodule? > > 0(3981) parse error (267,20-21): unknown command, missing loadmodule? > > 0(3981) parse error (281,24-25): unknown command, missing loadmodule? > > ERROR: bad config file (31 errors) > > > > > > ___________________________________________________________ > 1GB gratis, Antivirus y Antispam > Correo Yahoo!, el mejor correo web del mundo > http://correo.yahoo.com.ar > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

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FW: [Serusers] ser live database and mysql
by Richard C. Thompson 24 Jan '06

24 Jan '06

Frank Thanks for the reply I thought that setting "modparam("usrloc", "db_mode", 2)" to 1 did the same thing. ie it updates the table whenever anything changes. I had it set to 1 but the "location" sql database swelled to over 9000 and ser wouldn't restart until I truncated the table. My concern is that ser can't delete records in sql even though I checked the grants and privileges for user ser and they seem to be ok. Also serctl alias rm doesn't remove the aliases record from the table, again an issue with grants. But serctl alias add works. | Grants for ser@localhost | +--------------------------------------------------------------------------- --------+ | GRANT USAGE ON *.* TO 'ser'@'localhost' IDENTIFIED BY PASSWORD '*******************************' | | GRANT ALL PRIVILEGES ON `ser`.* TO 'ser'@'localhost' | +--------------------------------------------------------------------------- --------+ Got any ideas? Rick _____ From: Frank Fischer [mailto:frank.fischer@digitalnomads.ch] Sent: Tuesday, January 24, 2006 3:25 AM To: RThompson(a)vir2com.com Subject: RE: [Serusers] ser live database and mysql ser keeps the location table in memory for performance reasons and does synchronisation with the backend database sometimes. As far as i understand, this could lead to differences between in-memory data and in database-data. To avoid this situation and force SER to write all data to the database, you could use the experimental module "usrloc-cl". It works fine for me. - Frank _____ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Richard C. Thompson Sent: Monday, January 23, 2006 9:55 PM To: serusers(a)lists.iptel.org Subject: [Serusers] ser live database and mysql Hi all Hopefully someone can answer this. I am running ser with mysql and I have noticed that serctl moni shows a smaller number in "location" than the mysql "location" table. Eg serctl moni shows 404 in location mysql shows 556 in location I have modparam("usrloc", "db_mode", 2) even when I refresh the sql query many times over a long time, the tables never match. It seems that ser can't delete some of the entries Any help or hints would be appreciated Rick

1 0

[Users] trusting gateways through radius
by Jayesh Nambiar 24 Jan '06

24 Jan '06

Hi all, I have configured openser(1.0.0) to authenticate and account via freeradius. It is working properly and fine. But one problem that I am facing is trusting the gateways. I dont want to authenticate the gateways but trust them through their IP address. Is there a way I can do it in freeradius? If avp_radius can be used, pls explain where do i store the IPs that I trust in freeradius. If anyone can help in this regard it would be appreciated. Thanks a lot in advance. Jayesh. Send instant messages to your online friends http://in.messenger.yahoo.com

2 2

[Users] SER as loadbalancer and NAT
by Andreas Granig 24 Jan '06

24 Jan '06

Hi, A question to the NAT experts: I wonder if there are known side effects if I use fix_nated_contact() in combination with nat_flag in the registrar module instead of fix_nated_register() to save NATed contacts? What I want to do is checking for NAT on a stateless SER acting as load balancer, then forwarding it to a proxy in the LAN, like: # remove P-Nated-HF here if present... if(method == REGISTER && !search("^Contact:[ ]*\*") && client_nat_test("7")) { fix_nated_contact(); append_hf("P-Nated: 1\r\n"); force_rport(); } # forwarding to proxy using dispatcher module And in the proxy: modparam("registrar", "nat_flag", 1) # ... if(method == REGISTER) { if(is_present_hf("P-Nated")) setflag(1); save("location"); } # ... if(method == INVITE) { # a request for a local user... lookup("location"); setdsturi("sip:<address of loadbalancer>"); t_relay(); } Of course a mechanism has to be provided to let the loadbalancer or an external application NAT-ping the address provided in the rewritten contact. As far as I have seen, the difference is that this approach would save the NAT-adress of the contact in usrloc with an empty received-param, while fix_nated_register() saves the private adress of the contact with the NAT-adress in the received-param. I would assume (not verified yet), that in the case of using fix_nated_contact(), the NAT-address is set as r-uri after a lookup, with an empty dst-uri, which will have to be set to the load balancer's address to traverse the client-side NAT. In case of using fix_nated_register(), I assume the r-uri is set to the client's private address, and the received-address as dst-uri. Traversing the client's NAT wouldn't be that easy in this case because you would have to save the received-address somewhere in the message and forward the request to the load-balancer, which in turn has to forward it to the client's NAT address. So would this approach using fix_nated_contact() for REGISTERs make sense for using a SER-based load balancer? Or are there any limitations I haven't considered? Thanks, Andy

3 9

[Users] cancel method problem
by unplug 24 Jan '06

24 Jan '06

When UA1 makes a call to UA2, both ringing. Then UA1 cancels the call, a strange log message (from ngrep) will be got below. After that UA2 will ring 3 times itself. You can see in line 718, there is a cancel method. After that, there are many Record-Route, Via in the sip CANCEL method (line 763, line 1578 and many). Does anyone know what the problem is? http://meerkat.no-ip.com/openser/cancel.log

3 4

[Serusers] Help require to install SER in RH9
by Dinesh TJ 24 Jan '06

24 Jan '06

Hello All, Am a newbee to SER and interested to know how it works. I am tryin to install SER in RH9 box and facing some issues. If somebody can send the detailed step by step procedure on how to install then it will be really helpful for me. I am facing problem in Mysql part, first of all am not familiar with Mysql moreover i need to connect mysql with SER which is quite difficult for me. Please help me on this regard Regards Dinesh

1 0

[Serusers] ser live database and mysql
by Richard C. Thompson 23 Jan '06

23 Jan '06

Hi all Hopefully someone can answer this. I am running ser with mysql and I have noticed that serctl moni shows a smaller number in "location" than the mysql "location" table. Eg serctl moni shows 404 in location mysql shows 556 in location I have modparam("usrloc", "db_mode", 2) even when I refresh the sql query many times over a long time, the tables never match. It seems that ser can't delete some of the entries Any help or hints would be appreciated Rick

1 0

[Users] Call-specific AVP DBS storage for OpenSER
by Joachim Fabini 23 Jan '06

23 Jan '06

Hi all, Unfortunately the previous thread on this topic ended without a definitive statement or conclusion, probably because of the New Year holiday (btw, Happy New Year to all of you!). So here's another try: OpenSER is currently session stateful but not call stateful. This is valid also/especially for the AVP module: You can store AVPs on a per-contact base but (imho) _not_ specific to a call or dialog. The reason for this is that the AVP module does not allow/ implement the use of call-dependent variables as AVP names. Adding this feature to OpenSER doesn't require a change in interfaces (implementation effort is another topic ;). It's just about enabling the use of AVP or header values (like, e.g. call-id) as the _name_ of another AVP for DBS storage. I attach a specific example below. Can some of the developers and/or maintainers please comment and/or give some hints on the feasibility of this approach? Is it possible to add this feature to OpenSER? What effort is required to add this enhancement to the AVP module? Please come back if some details are unclear, I might have missed something important. Many thanks in advance for your reply and help, Best regards --Joachim PS: Example: The task is to process any SIP message that passes through our proxy. We must compute a unique ID specific to the call on the first message belonging to this call and store it in DBS. Whenever another message that is part of this call/dialog arrives, we retrieve the unique ID from the DBS and, e.g., append it as a specific header field value. The only way how this can be done is to use some call- specific AVP name (e.g. call-id), as we do not have any influence on the structure of the AVP value (so regexp- search based on AVP value does not help). # This is the AVP name that is required to retrieve # our specific, call-dependent AVP value from the # database whenever a message that belongs to this # call passes our OpenSER proxy. avp_printf("s:myTmpHeaderName", "$from/username$hdr(call-id)"); # This is the value we'd like to store. Could be also # only "$Ts", i.e., we can _not_ rely on regular # expression search for later retrieving the AVP # from the DBS based on its value. avp_printf("s:myTmpHeaderValue", "$hdr(call-id)-$Ts"); # This storing of the call-dependent info in DBS does # currently not work. OpenSER complains at start-up # about the variable AVP name/key on both of the # following lines. The former variant seems much more # flexible to me than the latter one. avp_db_store("s:myTmpHeaderName","s:myTmpHeaderValue"); avp_db_store("$hdr(call-id)","s:myTmpHeaderValue"); # This load obviously does also not work. avp_db_load("$hdr(call-id)","s:myTmpHeaderValue");

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[Serusers] Rtpproxy again
by Olivier Taylor 23 Jan '06

23 Jan '06

Another strange thing. When I call one of our users, it rings ok, but if I hangup and retry the same number, Ser seems not finding that user, if the user reset his phone, no problems, it rings again... Olivier

1 0

[Serusers] Push a specific entry in /etc/hosts
by Walter Luis 23 Jan '06

23 Jan '06

Hi, Friends, Is there a way (line in the server.conf) to push a especif entry in /etc/hosts to openvpn clients? I mean, something like: "command" domain.com 20.x.x.x Then, from the start to the finish of the openvpn, the client always knows that domain.com is 200.X.X.X and won't consult the dns. []'s -- Walter Luis

1 1

[Serusers] rtpproxy
by Olivier Taylor 23 Jan '06

23 Jan '06

Hi all, I have a ser(0.9.4/Freebsd) server well working, but in some cases, I have a strang behavior. I use rtpproxy for nat traversal,all my users can call pstn, but when we try to call one of our users from internal or from pstn, Ser sends many invites and callee never answer to these invite. I must conclude that they don't receive it... All the users having that problems are under a nated network. Any idea ? Olivier

1 0

[Serusers] Test
by sip 23 Jan '06

23 Jan '06

Having received zero messages on the ser list since the 20th, I thought I'd see if there's an issue with the list or if it's just been bizarrely quiet. N.

1 0

[Serusers] Asterisk and SER
by Jim McIver 23 Jan '06

23 Jan '06

Gentlemen, If I was using SER as a proxy server to deal with clients behind NATs, I can understand the concept - the sip proxy rewrites the header information to correct for the NATed client. Where I have a large "hole" in my understanding is concerning inbound calls from PSTN. Currently, these arrive at our Asterisk box which has no knowledge of the clever things SER should be doing to deal with the NATed clients. Can anyone please fill in the gaps in my understanding ? Better still.....can anyone let me have a look at their Asterisk and SER config files on a system that works ??? Jim -- --- www.yakara.com

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Re: [Serusers] Fwd: No RTP
by Voipers Portugal 23 Jan '06

23 Jan '06

SER does not handle RTP messages, only SIP messages. Jose Simoes On 1/16/06, samuel <samu60(a)gmail.com> wrote: > > www.onsip.org > > 2006/1/16, mohd suhaimi <see2belief(a)yahoo.com>: > > Hi, > > > > Any help on below is appreciated. > > > > Thanks > > > > --- mohd suhaimi <see2belief(a)yahoo.com> wrote: > > > > > Date: Thu, 12 Jan 2006 02:17:15 -0800 (PST) > > > From: mohd suhaimi <see2belief(a)yahoo.com> > > > Subject: No RTP > > > To: seruser <serusers(a)lists.iptel.org> > > > > > > Hi, > > > > > > when i make SIP to SIP calls thru my ser, no rtp > > > packets is processed. Especially when in a NATed > > > environment. > > > > > > Any idea? > > > > > > Thanks > > > > > > __________________________________________________ > > > Do You Yahoo!? > > > Tired of spam? Yahoo! Mail has the best spam > > > protection around > > > http://mail.yahoo.com > > > > > > > > > __________________________________________________ > > Do You Yahoo!? > > Tired of spam? Yahoo! Mail has the best spam protection around > > http://mail.yahoo.com > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

1 0

[Users] openserctl doesnt work
by martin＠campus-merseburg.de 23 Jan '06

23 Jan '06

Hi, when starting openserctl i get this message: root@sip:~ # /usr/local/sbin/openserctl /usr/local/sbin/openserctl: syntax error at line 1024: `GW_IP=$' unexpected OS is Solaris OpenSER Version: 1.0.0 CVS Bye

2 2

[Users] Asterisk & Openser
by Ronald Wiplinger 23 Jan '06

23 Jan '06

I want to use Openser to register all phones. If the phone is allowed (paid) than it should be able to reach a certain gateway (flat rate to certain numbers) Distinguished via mysql database. If the destination number is not in the flat rate, than the call should be forwarded to Astersik to handle the call (prepaid card) If features are requested, than should be also Asterisk handle it. Is there a starting point for this set-up?

2 1

[Users] connection between openser and mysql both in different machine
by unplug 23 Jan '06

23 Jan '06

How does openser connect to mysql which is not in the same machine (say openser is in 10.0.0.1 and mysql is in 10.0.0.2)? I have configured all mysql statement in the configuration file as "mysql://openser:openserrw@10.0.0.2/openser" but openser failed to start. Below is the error message from the log. Jan 21 14:25:51 owthkems01 /usr/local/sbin/openser[12298]: new_connection: Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (2) Jan 21 14:25:51 owthkems01 /usr/local/sbin/openser[12298]: ERROR:domain_db_init: cannot initialize database connection Jan 21 14:25:51 owthkems01 /usr/local/sbin/openser[12298]: ERROR: domain:child_init(): Unable to connect to the database Jan 21 14:25:51 owthkems01 /usr/local/sbin/openser[12298]: ERROR:init_mod_child: Error while initializing module domain, rank 1

2 2

[Users] question about avp
by unplug 23 Jan '06

23 Jan '06

Say I have a record in the table usr_preferences with username, attribute=function, value=voicemail. I use avp_db_load("$from/username", "s:function") to get the value. I can print out the value in xlog function by using $avp(s:function). How can I put that value in the header by using append_hf? As I found that it doens't work if I use append_hf("P-hint: $avp(s:function)\r\n"). In other words, how can I get back the value of the variable stored in avp for other functions processing?

3 3

RE: [Serusers] SER frontend
by Barta János 23 Jan '06

23 Jan '06

Hello! I'm again :-). If I use the next config in my ser.cfg, then the authentication works fine, but the presence does not. if (!www_authorize("domain.hu", "subscriber")) { forward( csps.domain.hu, 5060); break; }; But if I insert the save("location") line in my config file, then the presence info will be good, but the authentication will be wrong. if (!www_authorize("domain.hu", "subscriber")) { forward( csps.domain.hu, 5060); save("location"); break; }; Can I mix these two features? Thanks, Jani -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Barta János Sent: Friday, January 20, 2006 4:45 PM To: serusers(a)lists.iptel.org Subject: [Serusers] SER frontend Hello! I would like to make a frontend SIP perver. For this server will be only one task, to provide the presence information. The architecture: SIP client -> SER ->CSPS. I have a problem with the registration method. I'm using http digest auth. If I forward the authentication messages (to the CSPS), then the SIP client logs in automatically, and in this case I don't have the possibility to type the username/pwd. What is wrong? And I have a second question: Can I use W$ Messenger 5.1 presence under SER? (The 5.0 works fine, but the 5.1 does not.) This is my config file: sip:/etc/ser# cat ./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=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ alias=domain.hu alias=100.22.2.180 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" fifo_db_url="mysql://ser:heslo@localhost/ser" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/pa.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/exec.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- ####################### # -- usrloc params -- # ####################### /* 0 -- dont use mysql, 1 -- write_through, 2--write_back */ modparam("usrloc","db_mode",0) #modparam("usrloc","db_url","mysql://xxxxxxx:xxxx@localhost/ser") #modparam("usrloc", "use_domain", 1) #modparam("usrloc", "timer_interval", 10) ##################### # -- auth params -- # ##################### #modparam("auth_db","calculate_ha1", yes) #modparam("auth_db","db_url","mysql://xxxxx:xxxxx@localhost/ser") #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) ######################### # -- presence params -- # ######################### #modparam("pa", "db_url", "mysql://xxxxxxxx:xxxxxxxx@localhost/ser") modparam("pa", "default_expires",50) modparam("pa", "pa_domain", "domain.hu") # ------------------------- 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 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 (!t_relay()) { sl_reply_error(); }; break; }; if (uri==myself) { if (method=="REGISTER") { # if (!www_authorize("domain.hu", "subscriber")) { # forward( csps.domain.hu, 5060); # www_challenge("domain.hu", "0"); # break; # }; save("location"); break; }; lookup("aliases"); if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; forward( csps.domain.hu, 5060); break; }; sl_send_reply("404", "Not Found"); # append_hf("P-hint: usrloc applied\r\n"); } sip:/etc/ser# Thanks, Jani ________________________________________________________________ Harry Potter és a Félvér Herceg! Garantált szállítás a megjelenés napján! (február 10. ) Jegyezze elő most! http://www.bookline.hu/control/news?newsid=322&affiliate=frehp6kar1482

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[Serusers] AW: Serusers Digest, Vol 33, Issue 26, RE: Running mediaproxy
by "Töx, Reinhold" 23 Jan '06

23 Jan '06

>Gentlemen, >I tried running media proxy using the scripts given in the getting started document (putting the scripts into /etc/init.d/ as >shown in the document). >However, when I try to run it, I get the following error - >[root@itrhost2 init.d]# ./mediaproxy start >Starting SER MediaProxy server: mediaproxyTraceback (most recent call last): > File "/usr/local/mediaproxy/mediaproxy.py", line 21, in ? > from optparse import OptionParser >ImportError: No module named optparse >Traceback (most recent call last): > File "/usr/local/mediaproxy/proxydispatcher.py", line 35, in ? > from optparse import OptionParser >ImportError: No module named optparse >Any ideas what's wrong here ? Have you checked Python yet. Which version of Python do you use? Run 'python' from command line and type 'import optparse' to see what happens. If there is an error, you do not have the Python module 'optparse'. Upgrade to a newer version of Python or install the module by hand. Regards, Reinhold

1 0

[Users] substring function
by unplug 22 Jan '06

22 Jan '06

Does openser provide any substring function to get a substring within a string? For example, I have a string "1234abc5678" in avp s:gw, and I want to get "abc" from s:gw.

2 1

[Users] uac_replace_from function
by unplug 20 Jan '06

20 Jan '06

I am using uac_replace_from function as below. # replace only display and do not touch uri uac_replace_from("35418474",""); After running uac_replace_from function, I get the following message. U 203.193.4.242:5060 -> 203.193.4.226:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 203.193.4.226;branch=z9hG4bKaed7.46cc4101.0,SIP/2.0/UDP 10.0.0.52:5060;rport=5060;received=210.184.4.31;branch=z9hG4bKHNHpldVsKca6IYUS. From: "35418474"<sip:882754853589@ow01.owtel.com>;tag=sBl2gLH6O1myyqL5. To: "92588819" <sip:92588819@ow01.owtel.com>;tag=47C61DF4-26CD. Date: Fri, 06 Jan 2006 10:41:17 GMT. Call-ID: yTCQvS56ZCZXCHgt(a)10.0.0.52. Server: Cisco-SIPGateway/IOS-12.x. In the normal message, sip message, there is a space between display name and uri. ie. From: "35418474" <sip:882754853589@ow01.owtel.com> However, after I apply uac_replace_from function, there is a space there. How can I add a space such that it looks like a normal sip message? Does the space will affect the caller display?

3 10

[Users] centralized speed dial
by Reto Kortas 20 Jan '06

20 Jan '06

Hi! I am planning to set up a centralized speed dial service for a 2-layer proxy hierarchy. The design should look like these: PSTN | | UA --> LOCAL_SP --> MAIN-PROXY <-- LOCAL_SP <-- UA | | UA --> LOCAL_SP ------ | | PSTN ... and many many more local SIP Proxys (LOCAL_SP). Any branch has a local SIP-Proxy which functions also as a PSTN-Gateway and will route local calls, whether they are from inside or from the PSTN. The users are configured on the local SIP-Proxy, which will also register these user on a REGISTER-Request also on the centralized SIP-Proxy. The Speed dial database should contain internal as well as external (PSTN) numbers. The tricky thing is, that the users have to use their local PSTN Gateway, when they want to make off-net calls. The call flow must like these: UA -INVITE-> LOCAL_SP -INVITE-> MAIN-PROXY -> DB-Lookup for speed dial-alias MAIN-PROXY --301_RESPONSE--> LOCAL_SP --301_RESPONSE--> UA -> Setup new call with info from new contact header Does anybody now how to setup OpenSER, that he will execute a DB lookup for the user part of the R-URI and answers with a 301 response and the new address in the contact header? May I think the routing must look like this: Route[x] { NEW_URI = get_alias_for(1234); append_branch(NEW_URI); sl_send_reply("301" , "Number found"); } But how to set up and querie the database? Thx Reto

1 0

[Users] Question on re-writing From uri when ser forwards to Asterisk
by Barry Flanagan 20 Jan '06

20 Jan '06

Hi, I am using asterisk as PSTN gateway for users registered with OpenSER. Everything is working well for a single domain, but I will have problems with this when I add extra domains. I will have multiple domains using the OpenSER server, all using the same set of Asterisk servers for PSTN. My problem is that when user@domain1 gets sent to Asterisk, they are authenticated as just "user" (i.e. the domain portion is stripped off) and they get registered under the [user] sip name in Asterisk. Now, when I add the second domain, and user@domain2 goes to Asterisk, they also will appear to Asterisk to be [user] I think what I need to do is rewrite the From: URI to be user_domain1 and user_domain2 and then set them up in Asterisk as [user_domain1] and [user_domain2] so that they are unique So, my questions are: is this the correct/only way to do this, and if so, what do I need to do to achieve this? Many thanks. -- -Barry Flanagan

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[Users] only one call from duplicate register
by Ibrahim TUNALI 20 Jan '06

20 Jan '06

i know i can multiple register on a sip proxy with same account at same time from different locations. And both of them can call different numbers. But i want to only one client can call, other one will block during first one's conversation and second one will recive "not acceptable" response. is it possible and can you give a configuration example. regards, itunali

1 0

[Users] NAT+uac_replace_from
by unplug 20 Jan '06

20 Jan '06

Actually, I am replacing the username of the uri with the alias that stored in the database. In my configuration file, it is using mediaproxy for NAT function (features-callfwd.5.0.cfg from getting started). I also add the following codes in the very first of the route routine for alias replacing purpose. route { ... if (!has_totag() && method=="INVITE") { if (avp_db_load("$from/uri","s:alias")) { xlog("L_INFO","sip408: have alias - [$avp(s:alias)]\n"); uac_replace_from("anonymous","sip:$avp(s:alias)@$si"); } else { xlog("L_INFO","sip411: no alias\n"); }; }; ... } When I make a call from a phone to the PSTN phone, the caller drops the ring when the callee rings. Callee hangs up and callee rings again few seconds after. You can find an error message "SIP/2.0 481 Call Leg/Transaction Does Not Exist" in line 180. I wonder if there is any thing wrong with the above code or my concept is wrong. Please help. Below is the sip message. interface: any filter: (ip) and ( port 5060 ) # U 10.0.0.46:5060 -> 10.0.0.27:5060 INVITE sip:34163634@10.0.0.27 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.46:5060;branch=z9hG4bKeKunMgjqbhggzOxi. Max-Forwards: 70. User-Agent: Koncept KE10XX v4.32.05 00-09-45-0a-fc-2d. From: "9000" <sip:9000@10.0.0.27>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. Contact: <sip:9000@10.0.0.46:5060>. CSeq: 1 INVITE. Supported: 100rel, replaces. Content-Type: application/sdp. Content-Length: 286. . v=0. o=9000 58685689 12166570 IN IP4 10.0.0.46. s=SIP CALL. c=IN IP4 10.0.0.46. t=0 0. m=audio 6000 RTP/AVP 18 3 0 8 4 101. a=rtpmap:18 G729/8000. a=rtpmap:3 GSM/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:4 G723/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 10.0.0.27:5060 -> 10.0.0.46:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 10.0.0.46:5060;branch=z9hG4bKeKunMgjqbhggzOxi;rport=5060. From: "9000" <sip:9000@10.0.0.27>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. CSeq: 1 INVITE. Server: OpenSer (1.0.0 (i386/linux)). Content-Length: 0. Warning: 392 10.0.0.27:5060 "Noisy feedback tells: pid=3353 req_src_ip=10.0.0.46 req_src_port=5060 in_uri=sip:34163634@10.0.0.27 out_uri=sip:934163634@10.30.0.248 via_cnt==1". . # U 10.0.0.27:5060 -> 10.30.0.248:5060 INVITE sip:934163634@10.30.0.248 SIP/2.0. Record-Route: <sip:34163634@10.0.0.27:5060;nat=yes;ftag=l7473818dCOs9ZMG;lr=on>. Via: SIP/2.0/UDP 10.0.0.27;branch=z9hG4bK0d2f.18572a04.0. Via: SIP/2.0/UDP 10.0.0.46:5060;rport=5060;branch=z9hG4bKeKunMgjqbhggzOxi. Max-Forwards: 69. User-Agent: Koncept KE10XX v4.32.05 00-09-45-0a-fc-2d. From: "anonymous"<sip:36418473@10.0.0.46>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. Contact: <sip:9000@10.0.0.46:5060>. CSeq: 1 INVITE. Supported: 100rel, replaces. Content-Type: application/sdp. Content-Length: 287. . v=0. o=9000 58685689 12166570 IN IP4 10.0.0.46. s=SIP CALL. c=IN IP4 10.0.0.27. t=0 0. m=audio 35540 RTP/AVP 18 3 0 8 4 101. a=rtpmap:18 G729/8000. a=rtpmap:3 GSM/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:4 G723/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 10.30.0.248:5060 -> 10.0.0.27:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 10.0.0.27;branch=z9hG4bK0d2f.18572a04.0,SIP/2.0/UDP 10.0.0.46:5060;rport=5060;branch=z9hG4bKeKunMgjqbhggzOxi. From: "anonymous"<sip:36418473@10.0.0.46>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>;tag=1EBAD768-B8F. Date: Wed, 04 Jan 2006 09:15:35 GMT. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Allow-Events: telephone-event. Content-Length: 0. . # U 10.30.0.248:5060 -> 10.0.0.27:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 10.0.0.27;branch=z9hG4bK0d2f.18572a04.0,SIP/2.0/UDP 10.0.0.46:5060;rport=5060;branch=z9hG4bKeKunMgjqbhggzOxi. From: "anonymous"<sip:36418473@10.0.0.46>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>;tag=1EBAD768-B8F. Date: Wed, 04 Jan 2006 09:15:35 GMT. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Require: 100rel. RSeq: 1250. Allow-Events: telephone-event. Contact: <sip:934163634@10.30.0.248:5060>. Record-Route: <sip:34163634@10.0.0.27:5060;nat=yes;ftag=l7473818dCOs9ZMG;lr=on>. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 177. . v=0. o=CiscoSystemsSIP-GW-UserAgent 584 9780 IN IP4 10.30.0.248. s=SIP Call. c=IN IP4 10.30.0.248. t=0 0. m=audio 17340 RTP/AVP 18. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. # U 10.0.0.27:5060 -> 10.0.0.46:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 10.0.0.46:5060;rport=5060;branch=z9hG4bKeKunMgjqbhggzOxi. From: "9000" <sip:9000@10.0.0.27>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>;tag=1EBAD768-B8F. Date: Wed, 04 Jan 2006 09:15:35 GMT. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Require: 100rel. RSeq: 1250. Allow-Events: telephone-event. Contact: <sip:934163634@10.30.0.248:5060>. Record-Route: <sip:34163634@10.0.0.27:5060;nat=yes;ftag=l7473818dCOs9ZMG;lr=on>. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 175. . v=0. o=CiscoSystemsSIP-GW-UserAgent 584 9780 IN IP4 10.30.0.248. s=SIP Call. c=IN IP4 10.0.0.27. t=0 0. m=audio 35540 RTP/AVP 18. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. # U 10.0.0.46:5060 -> 10.0.0.27:5060 PRACK sip:934163634@10.30.0.248:5060 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.46:5060;branch=z9hG4bK8UeWxCfGrMJSO66f. Route: <sip:34163634@10.0.0.27:5060;nat=yes;ftag=l7473818dCOs9ZMG;lr=on>. Max-Forwards: 70. User-Agent: Koncept KE10XX v4.32.05 00-09-45-0a-fc-2d. From: "9000" <sip:9000@10.0.0.27>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>;tag=1EBAD768-B8F. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. Contact: <sip:9000@10.0.0.46:5060>. CSeq: 2 PRACK. RAck: 1250 1 INVITE. Content-Length: 0. . # U 10.0.0.27:5060 -> 10.30.0.248:5060 PRACK sip:934163634@10.30.0.248:5060 SIP/2.0. Record-Route: <sip:10.0.0.27;ftag=l7473818dCOs9ZMG;lr=on>. Via: SIP/2.0/UDP 10.0.0.27;branch=z9hG4bKdc2f.dbee15d1.0. Via: SIP/2.0/UDP 10.0.0.46:5060;branch=z9hG4bK8UeWxCfGrMJSO66f. Max-Forwards: 69. User-Agent: Koncept KE10XX v4.32.05 00-09-45-0a-fc-2d. From: "9000" <sip:9000@10.0.0.27>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>;tag=1EBAD768-B8F. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. Contact: <sip:9000@10.0.0.46:5060>. CSeq: 2 PRACK. RAck: 1250 1 INVITE. Content-Length: 0. . #----------------------------------line 180 -------------------------- U 10.30.0.248:5060 -> 10.0.0.27:5060 SIP/2.0 481 Call Leg/Transaction Does Not Exist. Via: SIP/2.0/UDP 10.0.0.27;branch=z9hG4bKdc2f.dbee15d1.0,SIP/2.0/UDP 10.0.0.46:5060;branch=z9hG4bK8UeWxCfGrMJSO66f. From: "9000" <sip:9000@10.0.0.27>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>;tag=1EBAD768-B8F. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. CSeq: 2 PRACK. Content-Length: 0. . # U 10.0.0.27:5060 -> 10.0.0.46:5060 SIP/2.0 481 Call Leg/Transaction Does Not Exist. Via: SIP/2.0/UDP 10.0.0.46:5060;branch=z9hG4bK8UeWxCfGrMJSO66f. From: "9000" <sip:9000@10.0.0.27>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>;tag=1EBAD768-B8F. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. CSeq: 2 PRACK. Content-Length: 0. . # U 10.30.0.248:5060 -> 10.0.0.27:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 10.0.0.27;branch=z9hG4bK0d2f.18572a04.0,SIP/2.0/UDP 10.0.0.46:5060;rport=5060;branch=z9hG4bKeKunMgjqbhggzOxi. From: "anonymous"<sip:36418473@10.0.0.46>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>;tag=1EBAD768-B8F. Date: Wed, 04 Jan 2006 09:15:35 GMT. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Require: 100rel. RSeq: 1250. Allow-Events: telephone-event. Contact: <sip:934163634@10.30.0.248:5060>. Record-Route: <sip:34163634@10.0.0.27:5060;nat=yes;ftag=l7473818dCOs9ZMG;lr=on>. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 177. . v=0. o=CiscoSystemsSIP-GW-UserAgent 584 9780 IN IP4 10.30.0.248. s=SIP Call. c=IN IP4 10.30.0.248. t=0 0. m=audio 17340 RTP/AVP 18. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes.

3 5

[Users] 481 - Call leg Transaction
by unplug 20 Jan '06

20 Jan '06

In what case will cause the following error response. Does it happen in call forwarding? 481 - Call Leg Transcation does Not Exist Do I need to handle it or just ignore it?

2 4

[Users] only one call from duplicate register
by Ibrahim TUNALI 19 Jan '06

19 Jan '06

i know i can multiple register on a sip proxy with same account at same time from different locations. And both of them can call different numbers. But i want to only one client can call, other one will block during first one's conversation and second one will recive "not acceptable" response. is it possible and can you give a configuration example. regards, itunali

1 0

[Users] textops regular expression matching = subst to replace from field
by Neill Wilkinson 18 Jan '06

18 Jan '06

Hi all, Looking to get a way to replace the From field in a REGISTER message to the forward on to another Registrar, as a third party registration I've tried something along the lines of: If (subs("/(^From:)([ ]*)/\1 sip:newaddress@new.server/I")) log(1,"Match!\n"); else log(1,"No match\n"); I basically need help with the Regex syntax and a good means of checking - is egrep or sed a good means of testing? Thanks..... Neill...;o) --- avast! Antivirus: Outbound message clean. Virus Database (VPS): 0603-2, 17/01/2006 Tested on: 1/18/2006 11:16:18 avast! - copyright (c) 1988-2004 ALWIL Software. http://www.avast.com

2 4

[Users] call forward
by unplug 18 Jan '06

18 Jan '06

I am using the sample configuration file called "features-callfwd.5.0.cfg" with some modification. UA dials a 8 digits number and if it is not in the location table, prefix('9') will be appended in the number and route to the PSTN gateway. According to the configuration below, route 6 is responsible for call forward. With appropriate records in usr_preferences, openser can perform direct forward and forward on busy. However, in foward on no answer, one of the party can't hear any voice from the other party. I think the flow is something like this. UA1 openser UA5 UA6 ------INVITE----> --------INVITE------------> ... <-------time out/no answer- ------------------INVITE------------------> ... <--------------RTP established----------------------------------> result: UA6 can hear UA1, but UA1 can't get any sound from UA6 Anyone can help if there is any mistable in my configuration file to fix the above problem. Thanks! ------------start configuration------------- # $Id: features-callfwd.cfg 15 2005-09-15 17:37:10Z /CN=Greger V. Teigre/emailAddress=greger(a)onsip.org $ debug=3 fork=yes log_stderror=no listen=203.193.16.226 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=yes rev_dns=no fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" #fifo_mode=0666 #fifo permissions can be changed here log_facility=LOG_LOCAL0 alias=01.ol.com 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/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/permissions.so" loadmodule "/usr/local/lib/openser/modules/acc.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/uac.so" loadmodule "/usr/local/lib/openser/modules/group.so" modparam("auth_db|permissions|uri_db|usrloc|acc|group", "db_url", "mysql://openser:openserrw@localhost/openser") 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) #minimize write back window - default is 60 seconds modparam("usrloc", "timer_interval", 10) #use deomain information for user #modparam("usrloc", "use_domain", 1) modparam("registrar", "nat_flag", 6) #The maximum expires value of a Contact modparam("registrar", "max_expires", 120) #all contacts will be ordered in descending modification time order modparam("registrar", "desc_time_order", 1) #modparam("registrar", "use_domain", 1) modparam("rr", "enable_full_lr", 1) #Timer which hits if no final reply for an INVITE modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("avpops", "avp_url", "mysql://openser:openserrw@localhost/openser") modparam("avpops", "avp_table", "usr_preferences") modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "db_flag", 1) modparam("uac", "from_restore_mode", "auto") route { xlog("L_INFO","---[$fU-MR]---\n"); # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { xlog("L_INFO","---[$fU-main(91)]: 483 Too Many Hops---\n"); sl_send_reply("483", "Too Many Hops"); return; }; if (msg:len > max_len) { xlog("L_INFO","---[$fU-main(97)]: 483 Message Overflow---\n"); sl_send_reply("513", "Message Overflow"); return; }; #----------------------------------------------------------------- # detect the anonymous incoming call and add anonymous to the uri #----------------------------------------------------------------- if (!has_totag() && (method=="INVITE"||method=="ACK"||method=="CANCEL"||method=="BYE")) { route(7); } #------------------------------------------------------------------ # Accoutnng Section #------------------------------------------------------------------ setflag(1); #t_relay(); # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("203.193.16.226:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { xlog("L_INFO","---[$fU-main(137)]: loose_route()---\n"); if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3")||search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- #domian module if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { xlog("L_INFO","---[$fU-main(159)]: 403 Forbidden---\n"); sl_send_reply("403", "Forbidden"); }; return; }; if (method=="CANCEL") { route(1); return; } else if (method=="INVITE") { route(3); return; } else if (method=="REGISTER") { route(2); return; } else if (method=="ACK") { route(1); return; }; lookup("aliases"); #xlog("L_INFO","SIP line 191: [$fu] to [$tu]\n"); if (uri!=myself) { route(4); route(1); return; }; if (!lookup("location")) { xlog("L_INFO","---[$fU-main(188)]: 404 User Not Found---\n"); sl_send_reply("404", "User Not Found"); return; }; route(1); } route[1] { xlog("L_INFO","---[$fU-R1]---\n"); # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; #xlog("L_INFO","SIP Request line 220: [$rm] from [$fu] to [$tu]\n"); } route[2] { xlog("L_INFO","---[$fU-R2]---\n"); # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); return; }; if (!check_to()) { xlog("L_INFO","---[$fU-route2(237)]: 401 Unauthorized---\n"); sl_send_reply("401", "Unauthorized"); return; }; #------------check if the account is activated ---------- if (!is_user_in("From", "active")) { xlog("L_INFO","---[$fU-route2(244)]: 402 Payment Required---\n"); sl_send_reply("402", "Payment Required"); #append_hf("P-hint: PAYMANT REQUIRED\r\n"); return; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { xlog("L_INFO","---[$fU-R3]---\n"); # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- #if (!allow_trusted()) { #if (!proxy_authorize("","subscriber")) { #proxy_challenge("","0"); #return; #} else if (!check_from()) { #sl_send_reply("403", "Use From=ID"); #return; #}; #consume_credentials(); #}; if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; #if (uri=~"^sip:1[0-9]{10}@") { #strip(1); #}; lookup("aliases"); #xlog("L_INFO","SIP line 292: [$fu] to [$tu]\n"); if (uri!=myself) { route(4); route(1); return; }; #if (uri=~"^sip:011[0-9]*@") { #route(4); #route(5); #return; #}; if (avp_db_load("$ruri/username", "s:callfwd")) { xlog("L_INFO","---[$fU-route3(305)]: destination uri: [$ru]---\n"); setflag(22); avp_pushto("$ruri", "s:callfwd"); xlog("L_INFO","---[$fU-route3(308)]: destination uri: [$ru]---\n"); route(6); return; }; if (!lookup("location")) { xlog("L_INFO","---[$fU]-(route3:315) destination uri: [$du]---\n"); if (uri=~"^sip:[0-9]*@") { xlog("L_INFO","---[$fU-route3(317)]: sip:(0-9*@)---\n"); prefix("9"); route(4); route(5); return; }; xlog("L_INFO","---[$fU-route3(324)]: 404 User Not Found---\n"); sl_send_reply("404", "User Not Found"); return; }; if (avp_db_load("$ruri/username", "s:fwdbusy")) { if (!avp_check("s:fwdbusy", "eq/$ruri/i")) { xlog("L_INFO","---[$fU-route3(331)]: setflag(26)---\n"); setflag(26); }; }; if (avp_db_load("$ruri/username", "s:fwdnoanswer")) { if (!avp_check("s:fwdnoanswer", "eq/$ruri/i")) { xlog("L_INFO","---[$fU-route3(338)]: setflag(27)---\n"); setflag(27); }; }; t_on_failure("1"); route(4); route(1); } route[4] { xlog("L_INFO","---[$fU-R4]---\n"); # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { xlog("L_INFO","---[$fU-route4(357)]: $ru---\n"); if (!isflagset(8)) { xlog("L_INFO","---[$fU-route4(359)]: $ru---\n"); setflag(8); use_media_proxy(); }; }; } route[5] { xlog("L_INFO","---[$fU-R5]---\n"); # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("203.193.16.242"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); t_on_failure("1"); route(4); route(1); } route[6] { xlog("L_INFO","---[$fU-R6]---\n"); # ------------------------------------------------------------------------ # Call Forwarding Reply Route Handler # # This must be done as a route block because sl_send_reply() cannot be # called from the failure_route block # ------------------------------------------------------------------------ #if (uri=~"^sip:1[0-9]{10}@") { #strip(1); #}; lookup("aliases"); if (!isflagset(22)) { append_branch(); }; if (uri!=myself) { xlog("L_INFO","---[$fU-route6(405)]: (uri!=myyself)---\n"); route(4); route(1); return; }; #if (uri=~"^sip:011[0-9]*@") { } #if (uri=~"^sip:[0-9]*@") { #xlog("L_INFO","---[$fU-route6(415)]: (app prefix(9))---\n"); #prefix("9"); #route(4); #route(5); #return; #}; if (!lookup("location")) { if (uri=~"^sip:[0-9]*@") { xlog("L_INFO","---[$fU-route6(425)]: (sip:(0-9*@)---\n"); route(4); route(1); return; }; xlog("L_INFO","---[$fU-route6(432)]: 404 User Not Found---\n"); sl_send_reply("404", "User Not Found"); }; route(4); route(1); } route[7] { xlog("L_INFO","---[$fU-R7]---\n"); #if (method=="INVITE"||method=="ACK"||method=="CANCEL"||method=="BYE") { #if (is_user_in("from","")) { #xlog("L_INFO","SIP is_user_in=true\n"); #uac_replace_from("anonymous","sip:anonymous@s01.ol.com"); #} #} if (avp_db_load("$from/username","s:alias")) { xlog("L_INFO","---[$fU-route7(450)]: have alias from aliases- [$avp(s:alias)]---\n"); #uac_replace_from("$avp(s:alias)","sip:$avp(s:alias)@$si"); #uac_replace_from("$avp(s:alias) " ,""); } else { xlog("L_INFO","---[$fU-route7(454)]: no alias from aliases---\n"); }; } onreply_route[1] { xlog("L_INFO","---[$fU-OR1]---\n"); if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } failure_route[1] { xlog("L_INFO","---[$fU-FR1]---\n"); if (t_check_status("487")) { return; }; if (isflagset(26) && t_check_status("486")) { if (avp_pushto("$ruri", "s:fwdbusy")) { xlog("L_INFO","---[$fU-failure_route1(505)]: fwdbusy---\n"); avp_delete("s:fwdbusy"); resetflag(26); route(6); return; }; }; if (isflagset(27) && t_check_status("408")) { if (avp_pushto("$ruri", "s:fwdnoanswer")) { xlog("L_INFO","---[$fU-failure_route1(515)]: fwdnoanswer---\n"); avp_delete("s:fwdnoanswer"); resetflag(27); route(6); return; }; }; end_media_session(); } -----------end of configuration----------------

1 0

[Users] How to check destination set after branch_route?
by Papadopoulos Georgios 18 Jan '06

18 Jan '06

Hello, I am trying to filter out requests going to some IPs. So, I have a branch_route block where I drop the branches that go to the IPs I want to block. After the t_on_branch command I call t_relay. This works fine if the callee has registered with at least one "good" IP. If the callee has registered with only one IP which I want to block then there is only one branch that is getting droped. The following t_relay results in a 500 Server error. Is there a way to check if all branches have been dropped? I tried using isdsturiset() but it did not work. Even if there is only one branch and this branch has been dropped, isdsturiset() returns true. Is this correct or am I wrong somewhere? thank you for any help 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

[Users] Sending stored messages
by Andréa Giordanna 18 Jan '06

18 Jan '06

Hi Daniel, Thanks for all the informations. Based on those informations, Loide Mara (lmvj(a)dcc.ufam.edu.br) and I have made changes on msilo module code to make it able to send messages at a time stored in the silo table. Applying the patch patch-msilo-060117.diff the msilo module becomes able to send messages at the time stored on the table. Following the installation instructions for msilo module in http://openser.org/docs/modules/1.0.x/msilo.html#AEN199, it is just necessary to add a field in the silo table: sen_time (integer). This field will store the sending time of the stored messages. For offline messages, sending time equals incoming time. It can be useful if users are interested in store messages to be sent to one or more different users at a given time. Reminders for meetings or special dates could be sent through it. Here is a patch if someone is interested in testing it. Any comments will be appreciated. Regards, Andrea Giordanna --------------------------------- Yahoo! doce lar. Faça do Yahoo! sua homepage. diff -urN msilo_orig/msilo.c msilo/msilo.c --- msilo_orig/msilo.c 2006-01-17 19:55:34.000000000 -0400 +++ msilo/msilo.c 2006-01-17 19:57:25.000000000 -0400 @@ -71,7 +71,7 @@ #include "msfuncs.h" #define MAX_DEL_KEYS 1 -#define NR_KEYS 9 +#define NR_KEYS 10 /* LabVoIP: old value 9 */ char *sc_mid = "mid"; /* 0 */ char *sc_from = "src_addr"; /* 1 */ @@ -82,6 +82,7 @@ char *sc_ctype = "ctype"; /* 6 */ char *sc_exp_time = "exp_time"; /* 7 */ char *sc_inc_time = "inc_time"; /* 8 */ +char *sc_sen_time = "sen_time"; /* LabVoIP: 9 */ #define SET_STR_VAL(_str, _res, _r, _c) \ if (RES_ROWS(_res)[_r].values[_c].nul == 0) \ @@ -144,6 +145,7 @@ void destroy(void); void m_clean_silo(unsigned int ticks, void *); +void m_send_ready(); /* LabVoIP */ /** TM callback function */ static void m_tm_callback( struct cell *t, int type, struct tmcb_params *ps); @@ -245,6 +247,8 @@ } register_timer( m_clean_silo, 0, ms_check_time); + register_timer( m_send_ready, 0, ms_check_time); /* LabVoIP */ + reg_addr.s = ms_registrar; reg_addr.len = (ms_registrar)?strlen(ms_registrar):0; @@ -553,6 +557,13 @@ db_vals[nr_keys].val.int_val = val; nr_keys++; + /* LabVoIP: add sending time: equals incoming time */ + db_keys[nr_keys] = sc_inc_time; + db_vals[nr_keys].type = DB_INT; + db_vals[nr_keys].nul = 0; + db_vals[nr_keys].val.int_val = val; + nr_keys++; + if (msilo_dbf.use_table(db_con, ms_db_table) < 0) { LOG(L_ERR, "MSILO:m_store: Error in use_table\n"); @@ -639,9 +650,9 @@ struct to_body to, *pto = NULL; db_key_t db_keys[2]; db_val_t db_vals[2]; - db_key_t db_cols[6]; + db_key_t db_cols[7]; db_res_t* db_res = NULL; - int i, db_no_cols = 6, db_no_keys = 2, mid, n; + int i, db_no_cols = 7, db_no_keys = 2, mid, n; char hdr_buf[1024], body_buf[1024]; struct sip_uri puri; @@ -658,6 +669,7 @@ db_cols[3]=sc_body; db_cols[4]=sc_ctype; db_cols[5]=sc_inc_time; + db_cols[6]=sc_sen_time; DBG("MSILO:m_dump: ------------ start ------------\n"); @@ -759,48 +771,62 @@ DBG("MSILO:m_dump: dumping [%d] messages for <%.*s>!!!\n", RES_ROW_N(db_res), pto->uri.len, pto->uri.s); + int cur_time = (int) time(NULL); + for(i = 0; i < RES_ROW_N(db_res); i++) { mid = RES_ROWS(db_res)[i].values[0].val.int_val; - if(msg_list_check_msg(ml, mid)) - { - DBG("MSILO:m_dump: message[%d] mid=%d already sent.\n", - i, mid); - continue; - } + + if (RES_ROWS(db_res)[i].values[6].val.int_val > cur_time) { + DBG("MSILO:m_send_ready: message[%d] mid=%d not sent. Not ready!!!\nCurrent time: %d - Sending time: %d\n", i, mid, cur_time, RES_ROWS(db_res)[i].values[6].val.int_val); + } else { + + DBG("MSILO:m_send_ready: message[%d] mid=%d sent. Ready!!!\nCurrent time: %d - Sending time: %d\n", i, mid, cur_time, RES_ROWS(db_res)[i].values[6].val.int_val); + + if(msg_list_check_msg(ml, mid)) + { + DBG("MSILO:m_dump: message[%d] mid=%d already sent.\n", i, mid); + + if (msg_list_get_flag(ml, mid) & MS_MSG_ERRO) { + DBG("MSILO:m_dump: message[%d] mid=%d is a message to an offline user, or was not sent successfully. We're gonna send it again.\n", i, mid); + } else { + DBG("MSILO:m_dump: message[%d] mid=%d was sent successfully. Next...\n", i, mid); + continue; + } + } - memset(str_vals, 0, 4*sizeof(str)); - SET_STR_VAL(str_vals[0], db_res, i, 1); /* from */ - SET_STR_VAL(str_vals[1], db_res, i, 2); /* to */ - SET_STR_VAL(str_vals[2], db_res, i, 3); /* body */ - SET_STR_VAL(str_vals[3], db_res, i, 4); /* ctype */ + memset(str_vals, 0, 4*sizeof(str)); + SET_STR_VAL(str_vals[0], db_res, i, 1); /* from */ + SET_STR_VAL(str_vals[1], db_res, i, 2); /* to */ + SET_STR_VAL(str_vals[2], db_res, i, 3); /* body */ + SET_STR_VAL(str_vals[3], db_res, i, 4); /* ctype */ - hdr_str.len = 1024; - if(m_build_headers(&hdr_str, str_vals[3] /*ctype*/, + hdr_str.len = 1024; + if(m_build_headers(&hdr_str, str_vals[3] /*ctype*/, str_vals[0]/*from*/) < 0) - { - DBG("MSILO:m_dump: headers building failed!!!\n"); - if (msilo_dbf.free_result(db_con, db_res) < 0) - DBG("MSILO:m_dump: Error while freeing result of" + { + DBG("MSILO:m_dump: headers building failed!!!\n"); + if (msilo_dbf.free_result(db_con, db_res) < 0) + DBG("MSILO:m_dump: Error while freeing result of" " query\n"); - msg_list_set_flag(ml, mid, MS_MSG_ERRO); - goto error; - } + msg_list_set_flag(ml, mid, MS_MSG_ERRO); + goto error; + } - DBG("MSILO:m_dump: msg [%d-%d] for: %.*s\n", i+1, mid, + DBG("MSILO:m_dump: msg [%d-%d] for: %.*s\n", i+1, mid, pto->uri.len, pto->uri.s); - /** sending using TM function: t_uac */ - body_str.len = 1024; - rtime = - (time_t)RES_ROWS(db_res)[i].values[5/*inc time*/].val.int_val; - n = m_build_body(&body_str, rtime, str_vals[2/*body*/]); - if(n<0) - DBG("MSILO:m_dump: sending simple body\n"); - else - DBG("MSILO:m_dump: sending composed body\n"); + /** sending using TM function: t_uac */ + body_str.len = 1024; + rtime = + (time_t)RES_ROWS(db_res)[i].values[5/*inc time*/].val.int_val; + n = m_build_body(&body_str, rtime, str_vals[2/*body*/]); + if(n<0) + DBG("MSILO:m_dump: sending simple body\n"); + else + DBG("MSILO:m_dump: sending composed body\n"); - tmb.t_request(&msg_type, /* Type of the message */ + tmb.t_request(&msg_type, /* Type of the message */ &pto->uri, /* Request-URI */ &str_vals[1], /* To */ &str_vals[0], /* From */ @@ -809,6 +835,7 @@ m_tm_callback, /* Callback function */ (void*)(long)mid /* Callback parameter */ ); + } } done: @@ -900,7 +927,7 @@ */ void m_tm_callback( struct cell *t, int type, struct tmcb_params *ps) { - DBG("MSILO:m_tm_callback: completed with status %d\n", ps->code); + DBG("MSILO:m_tm_callback: message %ld completed with status %d\n", (long)*ps->param, ps->code); if(!ps->param) { DBG("MSILO m_tm_callback: message id not received\n"); @@ -914,14 +941,155 @@ if(ps->code < 200 || ps->code >= 300) { DBG("MSILO:m_tm_callback: message <%ld> was not sent successfully\n", - (long)ps->param); - msg_list_set_flag(ml, (int)(long)ps->param, MS_MSG_ERRO); + (long)*ps->param); + msg_list_set_flag(ml, (int)(long)*ps->param, MS_MSG_ERRO); goto done; } - msg_list_set_flag(ml, (int)(long)ps->param, MS_MSG_DONE); + msg_list_set_flag(ml, (int)(long)*ps->param, MS_MSG_DONE); done: return; } +/* LabVoIP: m_send_ready + * It checks on database if there are any messages in time to be sent + * - giordanna + */ + +void m_send_ready () { + + struct to_body to, *pto = NULL; + db_key_t db_keys[2]; + db_key_t db_cols[7]; + db_res_t* db_res = NULL; + int i, db_no_cols = 7, mid, n; + char hdr_buf[1024], body_buf[1024]; + + str str_vals[4], hdr_str , body_str; + time_t rtime; + + /* init */ + db_keys[0]=sc_uri_user; + db_keys[1]=sc_uri_host; + db_cols[0]=sc_mid; + + db_cols[1]=sc_from; + db_cols[2]=sc_to; + db_cols[3]=sc_body; + db_cols[4]=sc_ctype; + db_cols[5]=sc_inc_time; + db_cols[6]=sc_sen_time; + + + DBG("MSILO:m_send_ready: ------------ Checking for messages to send! ------------\n"); + hdr_str.s=hdr_buf; + hdr_str.len=1024; + body_str.s=body_buf; + body_str.len=1024; + + + if (msilo_dbf.use_table(db_con, ms_db_table) < 0) + { + LOG(L_ERR, "MSILO:m_send_ready: Error in use_table\n"); + goto error; + } + + if((msilo_dbf.query(db_con,db_keys,NULL,NULL,db_cols,0, + db_no_cols, NULL,&db_res)!=0) || (RES_ROW_N(db_res) <= 0)) + { + DBG("MSILO:m_send_ready: no messages in database!\n"); + goto done; + } + + DBG("MSILO:m_send_ready: dumping [%d] messages!!!\n", RES_ROW_N(db_res)); + + + memset( &to , 0, sizeof(to) ); + + int cur_time = (int) time(NULL); + + /* Loop to send all the messages */ + for(i = 0; i < RES_ROW_N(db_res); i++) + { + // Getting message id + mid = RES_ROWS(db_res)[i].values[0].val.int_val; + + // If it is time to send... + if (RES_ROWS(db_res)[i].values[6].val.int_val > cur_time) { + DBG("MSILO:m_send_ready: message[%d] mid=%d not sent. Not ready!!!\nCurrent time: %d - Sending time: %d\n", i, mid, cur_time, RES_ROWS(db_res)[i].values[6].val.int_val); + } else { + + DBG("MSILO:m_send_ready: message[%d] mid=%d sent. Ready!!!\nCurrent time: %d - Sending time: %d\n", i, mid, cur_time, RES_ROWS(db_res)[i].values[6].val.int_val); + + // To field + to.uri.s = (char *) RES_ROWS(db_res)[i].values[2].val.string_val; + to.uri.len = strlen(RES_ROWS(db_res)[i].values[2].val.string_val); + if (to.uri.len <= 0) /* || to.error != PARSE_OK) */ + { + DBG("MSILO:m_send_ready: 'To' erro na atribui�o!\n"); + goto error; + } + pto = &to; + + if(msg_list_check_msg(ml, mid)) + { + DBG("MSILO:m_send_ready: message[%d] mid=%d already sent.\n", i, mid); + continue; + } + + memset(str_vals, 0, 4*sizeof(str)); + SET_STR_VAL(str_vals[0], db_res, i, 1); /* from */ + SET_STR_VAL(str_vals[1], db_res, i, 2); /* to */ + SET_STR_VAL(str_vals[2], db_res, i, 3); /* body */ + SET_STR_VAL(str_vals[3], db_res, i, 4); /* ctype */ + + hdr_str.len = 1024; + if(m_build_headers(&hdr_str, str_vals[3] /*ctype*/, + str_vals[0]/*from*/) < 0) + { + DBG("MSILO:m_send_ready: headers building failed!!!\n"); + if (msilo_dbf.free_result(db_con, db_res) < 0) + DBG("MSILO:m_send_ready: Error while freeing result of" + " query\n"); + msg_list_set_flag(ml, mid, MS_MSG_ERRO); + goto error; + } + + DBG("MSILO:m_send_ready: msg [%d-%d] for: %.*s\n", + i+1, mid, pto->uri.len, pto->uri.s); + + /** sending using TM function: t_uac */ + body_str.len = 1024; + rtime = + (time_t)RES_ROWS(db_res)[i].values[5/*inc time*/].val.int_val; + n = m_build_body(&body_str, rtime, str_vals[2/*body*/]); + if(n<0) + DBG("MSILO:m_send_ready: sending simple body\n"); + else + DBG("MSILO:m_send_ready: sending composed body\n"); + + tmb.t_request(&msg_type, /* Type of the message */ + &pto->uri, /* Request-URI */ + &str_vals[1], /* To */ + &str_vals[0], /* From */ + &hdr_str, /* Optional headers including CRLF */ + (n<0)?&str_vals[2]:&body_str, /* Message body */ + m_tm_callback, /* Callback function */ + (void*)(long) mid /* Callback parameter */ + ); + } + } + +done: + /** + * Free the result because we don't need it + * anymore + */ + if (db_res!=NULL && msilo_dbf.free_result(db_con, db_res) < 0) + DBG("MSILO:m_send_ready: Error while freeing result of query\n"); + + return; +error: + return; +} diff -urN msilo_orig/ms_msg_list.c msilo/ms_msg_list.c --- msilo_orig/ms_msg_list.c 2006-01-17 19:55:34.000000000 -0400 +++ msilo/ms_msg_list.c 2006-01-17 19:57:30.000000000 -0400 @@ -244,6 +244,37 @@ } /** + * LabVoIP: + * get flag for message with mid + */ +int msg_list_get_flag(msg_list ml, int mid) +{ + msg_list_el p0; + + if(!ml || mid==0) + goto errorx; + + lock_get(&ml->sem_sent); + + p0 = ml->lsent; + while(p0) + { + if(p0->msgid==mid) + { + DBG("MSILO: msg_list_get_flag: mid:%d fl:%d\n", p0->msgid, p0->flag); + goto done; + } + p0 = p0->next; + } + +done: + lock_release(&ml->sem_sent); + return p0->flag; +errorx: + return -1; +} + +/** * check if the messages from list were sent */ int msg_list_check(msg_list ml) @@ -262,9 +293,9 @@ p0 = ml->lsent; while(p0) { - if(p0->flag & MS_MSG_DONE || p0->flag & MS_MSG_ERRO) + if(p0->flag & MS_MSG_DONE) /*|| p0->flag & MS_MSG_ERRO)*/ /* Messages with error will be maintained in sent_list*/ { - DBG("MSILO: msg_list_check: mid:%d got reply\n", p0->msgid); + DBG("MSILO: msg_list_check: mid:%d got positive reply\n", p0->msgid); if(p0->prev) (p0->prev)->next = p0->next; else diff -urN msilo_orig/ms_msg_list.h msilo/ms_msg_list.h --- msilo_orig/ms_msg_list.h 2006-01-17 19:55:34.000000000 -0400 +++ msilo/ms_msg_list.h 2006-01-17 19:57:33.000000000 -0400 @@ -70,6 +70,7 @@ void msg_list_free(msg_list); int msg_list_check_msg(msg_list, int); int msg_list_set_flag(msg_list, int, int); +int msg_list_get_flag(msg_list ml, int mid); /* LabVoIP */ int msg_list_check(msg_list); msg_list_el msg_list_reset(msg_list);

2 1

[Users] NAT Problem?
by Craig Mariner 17 Jan '06

17 Jan '06

Hi List I have a problem with OpenSER. OpenSER has a public IP and is not behind a NAT. Two clients that are behind a NAT ring each other but there is no audio either way. Any advise please? Sencond scenario two clients both behind the same NAT. . Client1 calls the Client2, Client2 UA rings, Client2 answers the call but it does not connect. Client1 call does not get answered by Client2 and Client1 just hears the ringing tone. Any ideas? Best Regards -- Craig Mariner Director Wifi Specialist Ltd www.wifispecialist.co.uk www.freewhy-fi.net Building a Free Mesh Network Tel. 08703403427 email: wifibase(a)gmail.com

2 2

[Users] infinite register loop with eyeBeam when call forking one an SER proxy
by Dr. Rodney G. McDuff 17 Jan '06

17 Jan '06

I am using SER 0.9.4 as a sip proxy. In the proxy I have setup a permanent user location pointing to our cisco sip/PSTN gateway; the idea being that when I ring my sip address it rings both my PSTN phone and my sip client. When I use X-Lite it all works perfectly. X-Lite REGISTERs a couple of times and when I INVITE someone sip URI both phones rings. First phone that asnwers get the prize all others get a CANCEL. When I use eyebeam [eyeBeam release 3010n stamp 19039] all I get is a seemingly infinite loop of REGISTERs with incrementing Cseq. Below is a ngrep at the proxy. Has anyone seen this behavior and known if its a bug with eyeBeam or the SER config? T xxx.xxx.152.78:1758 -> xxx.xxx.2.40:5060 [AP] REGISTER sip:sip.domain SIP/2.0. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-4152d5791948a902-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:username@xxx.xxx.152.78:1758>. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 375 REGISTER. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Supported: eventlist. User-Agent: eyeBeam release 3010n stamp 19039. Authorization: Digest username="username",realm="sip.domain",nonce="43cc8e5a5cb47d54a7c640016623a782b1206e3b",uri="sip:sip.domain",response="cdae7a3787b5539f61c1b20087efc3e0",algorithm=MD5. Content-Length: 0. . # T xxx.xxx.2.40:5060 -> xxx.xxx.152.78:1758 [AP] SIP/2.0 100 Trying. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-4152d5791948a902-1--d87543-;rport=1758. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 375 REGISTER. Server: Sip EXpress router (0.9.4 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.2.40:5060 "Noisy feedback tells: pid=11879 req_src_ip=xxx.xxx.152.78 req_src_port=1758 in_uri=sip:sip.domain out_uri=sip:sip.domain via_cnt==1". . # T xxx.xxx.2.40:5060 -> xxx.xxx.152.78:1758 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-4152d5791948a902-1--d87543-;rport=1758. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=b5c61af9a6130f64fdf89dc761547723.a060. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 375 REGISTER. Contact: <sip:58220@sip.pstn.gw.ip>;q=1;expires=4294965840, <sip:username@xxx.xxx.152.57:5060>;expires=503;received="sip:xxx.xxx.152.57:5060", <sip:username@xxx.xxx.152.78:5060>;expires=1356, <sip:username@xxx.xxx.152.78:1758>;expires=3600. Server: Sip EXpress router (0.9.4 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.2.40:5060 "Noisy feedback tells: pid=11879 req_src_ip=xxx.xxx.152.78 req_src_port=1758 in_uri=sip:sip.domain out_uri=sip:sip.domain via_cnt==1". . ## T xxx.xxx.152.78:1758 -> xxx.xxx.2.40:5060 [AP] REGISTER sip:sip.domain SIP/2.0. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-3d42141dce2ee012-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:username@xxx.xxx.152.78:1758>. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 376 REGISTER. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Supported: eventlist. User-Agent: eyeBeam release 3010n stamp 19039. Authorization: Digest username="username",realm="sip.domain",nonce="43cc8e5a5cb47d54a7c640016623a782b1206e3b",uri="sip:sip.domain",response="cdae7a3787b5539f61c1b20087efc3e0",algorithm=MD5. Content-Length: 0. . # T xxx.xxx.2.40:5060 -> xxx.xxx.152.78:1758 [AP] SIP/2.0 100 Trying. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-3d42141dce2ee012-1--d87543-;rport=1758. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 376 REGISTER. Server: Sip EXpress router (0.9.4 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.2.40:5060 "Noisy feedback tells: pid=11879 req_src_ip=xxx.xxx.152.78 req_src_port=1758 in_uri=sip:sip.domain out_uri=sip:sip.domain via_cnt==1". . # T xxx.xxx.2.40:5060 -> xxx.xxx.152.78:1758 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-3d42141dce2ee012-1--d87543-;rport=1758. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=b5c61af9a6130f64fdf89dc761547723.80b3. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 376 REGISTER. Contact: <sip:58220@sip.pstn.gw.ip>;q=1;expires=4294965840, <sip:username@xxx.xxx.152.57:5060>;expires=503;received="sip:xxx.xxx.152.57:5060", <sip:username@xxx.xxx.152.78:5060>;expires=1356, <sip:username@xxx.xxx.152.78:1758>;expires=3600. Server: Sip EXpress router (0.9.4 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.2.40:5060 "Noisy feedback tells: pid=11879 req_src_ip=xxx.xxx.152.78 req_src_port=1758 in_uri=sip:sip.domain out_uri=sip:sip.domain via_cnt==1". . ## T xxx.xxx.152.78:1758 -> xxx.xxx.2.40:5060 [AP] REGISTER sip:sip.domain SIP/2.0. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-5861a63d14707c05-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:username@xxx.xxx.152.78:1758>. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 377 REGISTER. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Supported: eventlist. User-Agent: eyeBeam release 3010n stamp 19039. Authorization: Digest username="username",realm="sip.domain",nonce="43cc8e5a5cb47d54a7c640016623a782b1206e3b",uri="sip:sip.domain",response="cdae7a3787b5539f61c1b20087efc3e0",algorithm=MD5. Content-Length: 0. . # T xxx.xxx.2.40:5060 -> xxx.xxx.152.78:1758 [AP] SIP/2.0 100 Trying. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-5861a63d14707c05-1--d87543-;rport=1758. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 377 REGISTER. Server: Sip EXpress router (0.9.4 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.2.40:5060 "Noisy feedback tells: pid=11879 req_src_ip=xxx.xxx.152.78 req_src_port=1758 in_uri=sip:sip.domain out_uri=sip:sip.domain via_cnt==1". . # T xxx.xxx.2.40:5060 -> xxx.xxx.152.78:1758 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-5861a63d14707c05-1--d87543-;rport=1758. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=b5c61af9a6130f64fdf89dc761547723.2412. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 377 REGISTER. Contact: <sip:58220@sip.pstn.gw.ip>;q=1;expires=4294965839, <sip:username@xxx.xxx.152.57:5060>;expires=502;received="sip:xxx.xxx.152.57:5060", <sip:username@xxx.xxx.152.78:5060>;expires=1355, <sip:username@xxx.xxx.152.78:1758>;expires=3600. Server: Sip EXpress router (0.9.4 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.2.40:5060 "Noisy feedback tells: pid=11879 req_src_ip=xxx.xxx.152.78 req_src_port=1758 in_uri=sip:sip.domain out_uri=sip:sip.domain via_cnt==1". . ## U xxx.xxx.152.78:5060 -> xxx.xxx.2.40:5060 . # T xxx.xxx.152.78:1758 -> xxx.xxx.2.40:5060 [AP] REGISTER sip:sip.domain SIP/2.0. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-4675e1000b2e3f40-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:username@xxx.xxx.152.78:1758>. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 378 REGISTER. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Supported: eventlist. User-Agent: eyeBeam release 3010n stamp 19039. Authorization: Digest username="username",realm="sip.domain",nonce="43cc8e5a5cb47d54a7c640016623a782b1206e3b",uri="sip:sip.domain",response="cdae7a3787b5539f61c1b20087efc3e0",algorithm=MD5. Content-Length: 0. . # T xxx.xxx.2.40:5060 -> xxx.xxx.152.78:1758 [AP] SIP/2.0 100 Trying. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-4675e1000b2e3f40-1--d87543-;rport=1758. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 378 REGISTER. Server: Sip EXpress router (0.9.4 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.2.40:5060 "Noisy feedback tells: pid=11879 req_src_ip=xxx.xxx.152.78 req_src_port=1758 in_uri=sip:sip.domain out_uri=sip:sip.domain via_cnt==1". . # T xxx.xxx.2.40:5060 -> xxx.xxx.152.78:1758 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-4675e1000b2e3f40-1--d87543-;rport=1758. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=b5c61af9a6130f64fdf89dc761547723.9461. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 378 REGISTER. Contact: <sip:58220@sip.pstn.gw.ip>;q=1;expires=4294965839, <sip:username@xxx.xxx.152.57:5060>;expires=502;received="sip:xxx.xxx.152.57:5060", <sip:username@xxx.xxx.152.78:5060>;expires=1355, <sip:username@xxx.xxx.152.78:1758>;expires=3600. Server: Sip EXpress router (0.9.4 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.2.40:5060 "Noisy feedback tells: pid=11879 req_src_ip=xxx.xxx.152.78 req_src_port=1758 in_uri=sip:sip.domain out_uri=sip:sip.domain via_cnt==1". . ## T xxx.xxx.152.78:1758 -> xxx.xxx.2.40:5060 [AP] REGISTER sip:sip.domain SIP/2.0. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-5f45fd45452d0a56-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:username@xxx.xxx.152.78:1758>. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 379 REGISTER. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Supported: eventlist. User-Agent: eyeBeam release 3010n stamp 19039. Authorization: Digest username="username",realm="sip.domain",nonce="43cc8e5a5cb47d54a7c640016623a782b1206e3b",uri="sip:sip.domain",response="cdae7a3787b5539f61c1b20087efc3e0",algorithm=MD5. Content-Length: 0. . # T xxx.xxx.2.40:5060 -> xxx.xxx.152.78:1758 [AP] SIP/2.0 100 Trying. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-5f45fd45452d0a56-1--d87543-;rport=1758. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 379 REGISTER. Server: Sip EXpress router (0.9.4 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.2.40:5060 "Noisy feedback tells: pid=11879 req_src_ip=xxx.xxx.152.78 req_src_port=1758 in_uri=sip:sip.domain out_uri=sip:sip.domain via_cnt==1". . # T xxx.xxx.2.40:5060 -> xxx.xxx.152.78:1758 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-5f45fd45452d0a56-1--d87543-;rport=1758. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=b5c61af9a6130f64fdf89dc761547723.7aca. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 379 REGISTER. Contact: <sip:58220@sip.pstn.gw.ip>;q=1;expires=4294965839, <sip:username@xxx.xxx.152.57:5060>;expires=502;received="sip:xxx.xxx.152.57:5060", <sip:username@xxx.xxx.152.78:5060>;expires=1355, <sip:username@xxx.xxx.152.78:1758>;expires=3600. Server: Sip EXpress router (0.9.4 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.2.40:5060 "Noisy feedback tells: pid=11879 req_src_ip=xxx.xxx.152.78 req_src_port=1758 in_uri=sip:sip.domain out_uri=sip:sip.domain via_cnt==1". . ## T xxx.xxx.152.78:1758 -> xxx.xxx.2.40:5060 [AP] REGISTER sip:sip.domain SIP/2.0. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-277dc139635ed406-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:username@xxx.xxx.152.78:1758>. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 380 REGISTER. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Supported: eventlist. User-Agent: eyeBeam release 3010n stamp 19039. Authorization: Digest username="username",realm="sip.domain",nonce="43cc8e5a5cb47d54a7c640016623a782b1206e3b",uri="sip:sip.domain",response="cdae7a3787b5539f61c1b20087efc3e0",algorithm=MD5. Content-Length: 0. . # T xxx.xxx.2.40:5060 -> xxx.xxx.152.78:1758 [AP] SIP/2.0 100 Trying. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-277dc139635ed406-1--d87543-;rport=1758. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 380 REGISTER. Server: Sip EXpress router (0.9.4 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.2.40:5060 "Noisy feedback tells: pid=11879 req_src_ip=xxx.xxx.152.78 req_src_port=1758 in_uri=sip:sip.domain out_uri=sip:sip.domain via_cnt==1". . # T xxx.xxx.2.40:5060 -> xxx.xxx.152.78:1758 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-277dc139635ed406-1--d87543-;rport=1758. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=b5c61af9a6130f64fdf89dc761547723.5668. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 380 REGISTER. Contact: <sip:58220@sip.pstn.gw.ip>;q=1;expires=4294965839, <sip:username@xxx.xxx.152.57:5060>;expires=502;received="sip:xxx.xxx.152.57:5060", <sip:username@xxx.xxx.152.78:5060>;expires=1355, <sip:username@xxx.xxx.152.78:1758>;expires=3600. Server: Sip EXpress router (0.9.4 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.2.40:5060 "Noisy feedback tells: pid=11879 req_src_ip=xxx.xxx.152.78 req_src_port=1758 in_uri=sip:sip.domain out_uri=sip:sip.domain via_cnt==1". . ## T xxx.xxx.152.78:1758 -> xxx.xxx.2.40:5060 [AP] REGISTER sip:sip.domain SIP/2.0. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-3e3eaf4bb77a4835-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:username@xxx.xxx.152.78:1758>. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 381 REGISTER. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Supported: eventlist. User-Agent: eyeBeam release 3010n stamp 19039. Authorization: Digest username="username",realm="sip.domain",nonce="43cc8e5a5cb47d54a7c640016623a782b1206e3b",uri="sip:sip.domain",response="cdae7a3787b5539f61c1b20087efc3e0",algorithm=MD5. Content-Length: 0. . # T xxx.xxx.2.40:5060 -> xxx.xxx.152.78:1758 [AP] SIP/2.0 100 Trying. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-3e3eaf4bb77a4835-1--d87543-;rport=1758. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 381 REGISTER. Server: Sip EXpress router (0.9.4 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.2.40:5060 "Noisy feedback tells: pid=11879 req_src_ip=xxx.xxx.152.78 req_src_port=1758 in_uri=sip:sip.domain out_uri=sip:sip.domain via_cnt==1". . # T xxx.xxx.2.40:5060 -> xxx.xxx.152.78:1758 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-3e3eaf4bb77a4835-1--d87543-;rport=1758. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=b5c61af9a6130f64fdf89dc761547723.641b. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 381 REGISTER. Contact: <sip:58220@sip.pstn.gw.ip>;q=1;expires=4294965839, <sip:username@xxx.xxx.152.57:5060>;expires=502;received="sip:xxx.xxx.152.57:5060", <sip:username@xxx.xxx.152.78:5060>;expires=1355, <sip:username@xxx.xxx.152.78:1758>;expires=3600. Server: Sip EXpress router (0.9.4 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.2.40:5060 "Noisy feedback tells: pid=11879 req_src_ip=xxx.xxx.152.78 req_src_port=1758 in_uri=sip:sip.domain out_uri=sip:sip.domain via_cnt==1". . -- Dr. Rodney G. McDuff |Ex ignorantia ad sapientiam Manager, Strategic Technologies Group| Ex luce ad tenebras Information Technology Services | The University of Queensland | EMAIL: mcduff(a)its.uq.edu.au | TELEPHONE: +61 7 3365 8220 |

2 1

[Users] requirement of using mediaproxy
by unplug 17 Jan '06

17 Jan '06

I am using the features-callfwd.cfg script with mediaproxy in a testing server. I can make call directly from phone to phone and forward call. However, it is failed to forward call on no answer and forward call on busy. In my testing server, it only has a network interface. I wonder if it is the cause of the problem. Does it need to have 2 network interfaces (one for external and one for internal) for those forward features work?

2 1

[Users] openser+freeradius
by Henri Keski-Sikkilä 16 Jan '06

16 Jan '06

I installed freeradius. When I tried to register, syslog message is: Jan 14 20:39:36 localhost /usr/sbin/openser[1280]: rc_get_seqnbr: fscanf failure: /var/run/radius.seq Jan 14 20:39:38 localhost /usr/sbin/openser[1280]: ERROR:auth_radius:radius_authorize_sterman: rc_auth failed Any ideas? Henri

3 3

[Users] database description
by Daniel-Constantin Mierla 16 Jan '06

16 Jan '06

Hello, courtesy of Norman Brandinger, there is now a new tutorial describing the tables in OpenSER Database. http://openser.org/docs/OpenSER_Database_Structure.html The dokuwiki page which was create to collect users' contributions in this respect will point to this tutorial. If anyone wants to contribute with more description, please sent email to devel(a)openser.org or add the text in the dokiwiki page. http://openser.org/dokuwiki/doku.php?id=database_description Cheers, Daniel

1 0

[Users] How to use the failover support in the dispatcher module?
by Wolfgang Hottgenroth 16 Jan '06

16 Jan '06

Hi, on 18. 12. 2005 Daniel-Constantin Mierla accounted on this list that the dispatcher module now has a kind of failover support. I was really curious for this feature, since I had to config openser to support both load-balancing and failover. Failover works fine: if one PSTNGW is not available, the next one is tried, until one works. But unfortunately, for negative status reports (like 'busy'), the failure_route is also entered and thereby also the failover mechanism is triggered, with the result that a 'busy' is signalled to the A-side as 'no more gws'. (I've tried different mechanisms for ds_select_domain, also I tried with and without the append_branch in the failure_route block.) Any hints for me? Thank you very much! Wolfgang debug=3 fork=yes log_stderror=yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" # ------------------ module loading ---------------------------------- loadmodule "/opt/openser/lib/openser/modules/sl.so" loadmodule "/opt/openser/lib/openser/modules/tm.so" loadmodule "/opt/openser/lib/openser/modules/rr.so" loadmodule "/opt/openser/lib/openser/modules/maxfwd.so" loadmodule "/opt/openser/lib/openser/modules/usrloc.so" loadmodule "/opt/openser/lib/openser/modules/registrar.so" loadmodule "/opt/openser/lib/openser/modules/textops.so" loadmodule "/opt/openser/lib/openser/modules/xlog.so" loadmodule "/opt/openser/lib/openser/modules/dispatcher.so" # ----------------- setting module-specific parameters --------------- modparam("rr", "enable_full_lr", 1) modparam("xlog", "buf_size", 8192) modparam("dispatcher", "list_file", "/opt/openser/etc/openser/dispatcher.list") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 3) modparam("tm", "fr_timer", 5) # ------------------------- request routing logic ------------------- # main routing logic route[0] { xlog("L_NOTICE", "----> before route 1 <----\n"); route(1); xlog("L_NOTICE", "----> after route 1 <----\n"); } route[1] { if (loose_route()) { xlog("L_NOTICE", "*** loose routing enforced: $rm, $ru, $si, $du\n"); t_relay(); return; } record_route(); xlog("L_NOTICE", "REQUEST: $rm, $ru, $si, $du\n"); if (method=="REGISTER") { xlog("L_NOTICE", "*** register request from $si\n"); sl_send_reply("503", "No registrar service here"); return; } if (method=="INVITE" && ! uri=~"^sip:\+[1-9][0-9]+@") { xlog("L_NOTICE", "*** illegal sip uri: $ru\n"); sl_send_reply("484", "International number required"); return; } # --- customer specific part --------------------------------------- if ( src_ip==62.191.185.122 ) { xlog("L_NOTICE", "*** $rm from laptop-who, $si\n"); append_hf("P-Hint: from laptop-who\n"); append_hf("Remote-Party-ID: <sip:+492319721231@62.191.185.47>;party=calling;screen=yes;privacy=off\n"); if (method=="INVITE") { strip(1); prefix("+0000666666"); } ds_select_domain("00", "4"); if (method=="INVITE") { t_on_failure("2"); } xlog("L_NOTICE", "----> before t_relay <----\n"); t_relay(); xlog("L_NOTICE", "----> after t_relay <----\n"); return; } if ( src_ip==62.191.185.29 ) { xlog("L_NOTICE", "*** $rm from laptop-aholt, $si\n"); append_hf("P-Hint: from laptop-aholt\n"); append_hf("Remote-Party-ID: <sip:+492319721231@62.191.185.47>;party=calling;screen=yes;privacy=off\n"); if (method=="INVITE") { strip(1); prefix("+0100555555"); } ds_select_domain("01", "4"); if (method=="INVITE") { t_on_failure("2"); } xlog("L_NOTICE", "----> before t_relay <----\n"); t_relay(); xlog("L_NOTICE", "----> after t_relay <----\n"); return; } if ( src_ip==62.191.185.53 ) { xlog("L_NOTICE", "*** $rm from ipphone-who, $si\n"); append_hf("P-Hint: from ipphone-who\n"); append_hf("Remote-Party-ID: <sip:+492319721231@62.191.185.47>;party=calling;screen=yes;privacy=off\n"); if (method=="INVITE") { strip(1); prefix("+0100555555"); } ds_select_domain("01", "4"); if (method=="INVITE") { t_on_failure("2"); } xlog("L_NOTICE", "----> before t_relay <----\n"); t_relay(); xlog("L_NOTICE", "----> after t_relay <----\n"); return; } if ( src_ip==127.0.2.1 || src_ip==127.0.2.2 ) { xlog("L_NOTICE", "*** $rm from cust3, $si\n"); append_hf("P-Hint: from cust3\n"); append_hf("Remote-Party-ID: <sip:+492319721231@62.191.185.47>;party=calling;screen=yes;privacy=off\n"); if (method=="INVITE") { strip(1); prefix("+0200000003"); } ds_select_domain("02", "4"); if (method=="INVITE") { t_on_failure("2"); } xlog("L_NOTICE", "----> before t_relay <----\n"); t_relay(); xlog("L_NOTICE", "----> after t_relay <----\n"); return; } # ------------------------------------------------------------------ if (method=="INVITE") { xlog("L_NOTICE", "*** reject invite from $si\n"); t_reply("403", "Forbidden"); return; } xlog("L_NOTICE", "*** $rm from $si\n"); t_relay(); } failure_route[2] { xlog("L_NOTICE", "*** in failure_route 1: rc: $rc, rr: $rr, rs: $rs\n"); if (! ds_next_domain()) { t_reply("503", "Service unavailable: no more gws"); return; } else { append_branch(); t_on_failure("2"); t_relay(); return; } } # $Id: main.m4,v 1.3 2006/01/12 15:27:42 whottgen.uu.net Exp $ debug=3 fork=yes log_stderror=yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" # ------------------ module loading ---------------------------------- loadmodule "/opt/openser/lib/openser/modules/sl.so" loadmodule "/opt/openser/lib/openser/modules/tm.so" loadmodule "/opt/openser/lib/openser/modules/rr.so" loadmodule "/opt/openser/lib/openser/modules/maxfwd.so" loadmodule "/opt/openser/lib/openser/modules/usrloc.so" loadmodule "/opt/openser/lib/openser/modules/registrar.so" loadmodule "/opt/openser/lib/openser/modules/textops.so" loadmodule "/opt/openser/lib/openser/modules/xlog.so" loadmodule "/opt/openser/lib/openser/modules/dispatcher.so" # ----------------- setting module-specific parameters --------------- modparam("rr", "enable_full_lr", 1) modparam("xlog", "buf_size", 8192) modparam("dispatcher", "list_file", "/opt/openser/etc/openser/dispatcher.list") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 3) modparam("tm", "fr_timer", 5) # ------------------------- request routing logic ------------------- # main routing logic route[0] { xlog("L_NOTICE", "----> before route 1 <----\n"); route(1); xlog("L_NOTICE", "----> after route 1 <----\n"); } route[1] { if (loose_route()) { xlog("L_NOTICE", "*** loose routing enforced: $rm, $ru, $si, $du\n"); t_relay(); return; } record_route(); xlog("L_NOTICE", "REQUEST: $rm, $ru, $si, $du\n"); if (method=="REGISTER") { xlog("L_NOTICE", "*** register request from $si\n"); sl_send_reply("503", "No registrar service here"); return; } if (method=="INVITE" && ! uri=~"^sip:\+[1-9][0-9]+@") { xlog("L_NOTICE", "*** illegal sip uri: $ru\n"); sl_send_reply("484", "International number required"); return; } # --- customer specific part --------------------------------------- if ( src_ip==62.191.185.122 ) { xlog("L_NOTICE", "*** $rm from laptop-who, $si\n"); append_hf("P-Hint: from laptop-who\n"); append_hf("Remote-Party-ID: <sip:+492319721231@62.191.185.47>;party=calling;screen=yes;privacy=off\n"); if (method=="INVITE") { strip(1); prefix("+0000666666"); } ds_select_domain("00", "4"); if (method=="INVITE") { t_on_failure("2"); } xlog("L_NOTICE", "----> before t_relay <----\n"); t_relay(); xlog("L_NOTICE", "----> after t_relay <----\n"); return; } if ( src_ip==62.191.185.29 ) { xlog("L_NOTICE", "*** $rm from laptop-aholt, $si\n"); append_hf("P-Hint: from laptop-aholt\n"); append_hf("Remote-Party-ID: <sip:+492319721231@62.191.185.47>;party=calling;screen=yes;privacy=off\n"); if (method=="INVITE") { strip(1); prefix("+0100555555"); } ds_select_domain("01", "4"); if (method=="INVITE") { t_on_failure("2"); } xlog("L_NOTICE", "----> before t_relay <----\n"); t_relay(); xlog("L_NOTICE", "----> after t_relay <----\n"); return; } if ( src_ip==62.191.185.53 ) { xlog("L_NOTICE", "*** $rm from ipphone-who, $si\n"); append_hf("P-Hint: from ipphone-who\n"); append_hf("Remote-Party-ID: <sip:+492319721231@62.191.185.47>;party=calling;screen=yes;privacy=off\n"); if (method=="INVITE") { strip(1); prefix("+0100555555"); } ds_select_domain("01", "4"); if (method=="INVITE") { t_on_failure("2"); } xlog("L_NOTICE", "----> before t_relay <----\n"); t_relay(); xlog("L_NOTICE", "----> after t_relay <----\n"); return; } if ( src_ip==127.0.2.1 || src_ip==127.0.2.2 ) { xlog("L_NOTICE", "*** $rm from cust3, $si\n"); append_hf("P-Hint: from cust3\n"); append_hf("Remote-Party-ID: <sip:+492319721231@62.191.185.47>;party=calling;screen=yes;privacy=off\n"); if (method=="INVITE") { strip(1); prefix("+0200000003"); } ds_select_domain("02", "4"); if (method=="INVITE") { t_on_failure("2"); } xlog("L_NOTICE", "----> before t_relay <----\n"); t_relay(); xlog("L_NOTICE", "----> after t_relay <----\n"); return; } # ------------------------------------------------------------------ if (method=="INVITE") { xlog("L_NOTICE", "*** reject invite from $si\n"); t_reply("403", "Forbidden"); return; } xlog("L_NOTICE", "*** $rm from $si\n"); t_relay(); } failure_route[2] { xlog("L_NOTICE", "*** in failure_route 1: rc: $rc, rr: $rr, rs: $rs\n"); if (! ds_next_domain()) { t_reply("503", "Service unavailable: no more gws"); return; } else { append_branch(); t_on_failure("2"); t_relay(); return; } } main.cfg: main.mc m4/main.m4 m4/subs.m4 m4 -I ./m4 $< > $@ 00 sip:139.4.200.6:5060 01 sip:139.4.200.6:5060 include(`subs.m4') define(`_PKG_PATH_', `/opt/openser') define(`_INST_NAME_', `MAIN') define(`_DEBUG_LEVEL_', `3') CUST_BLOCK(`62.191.185.122', `laptop-who', `00', `0', `0666666') CUST_BLOCK(`62.191.185.29', `laptop-aholt', `01', `0', `0555555') CUST_BLOCK(`62.191.185.53', `ipphone-who', `01', `0', `0555555') CUST_BLOCK(`127.0.2.1,127.0.2.2', `cust3', `02', `0', `0000003') include(`main.m4') dnl dnl Main part of the SER configuration file dnl dnl dnl # $Id: main.m4,v 1.4 2006/01/13 14:29:06 whottgen.uu.net Exp $ divert(-1) ifdef(`_PKG_PATH_', `', ` errprint(`*** Package path not defined *** ')m4exit(1)') ifdef(`_INST_NAME_', `', ` errprint(`*** Instance name not defined *** ')m4exit(1)') define(`_MODULE_PATH_', _PKG_PATH_`/lib/openser/modules') ifdef(`_DEBUG_LEVEL_', `', ` define(_DEBUG_LEVEL_, `9')') divert debug=_DEBUG_LEVEL_ fork=yes log_stderror=yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" # ------------------ module loading ---------------------------------- loadmodule "_MODULE_PATH_/sl.so" loadmodule "_MODULE_PATH_/tm.so" loadmodule "_MODULE_PATH_/rr.so" loadmodule "_MODULE_PATH_/maxfwd.so" loadmodule "_MODULE_PATH_/usrloc.so" loadmodule "_MODULE_PATH_/registrar.so" loadmodule "_MODULE_PATH_/textops.so" loadmodule "_MODULE_PATH_/xlog.so" loadmodule "_MODULE_PATH_/dispatcher.so" # ----------------- setting module-specific parameters --------------- modparam("rr", "enable_full_lr", 1) modparam("xlog", "buf_size", 8192) modparam("dispatcher", "list_file", "/opt/openser/etc/openser/dispatcher.list") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 3) modparam("tm", "fr_timer", 5) # ------------------------- request routing logic ------------------- # main routing logic route[0] { xlog("L_NOTICE", "----> before route 1 <----\n"); route(1); xlog("L_NOTICE", "----> after route 1 <----\n"); } route[1] { if (loose_route()) { xlog("L_NOTICE", "*** loose routing enforced: $rm, $ru, $si, $du\n"); t_relay(); return; } record_route(); xlog("L_NOTICE", "REQUEST: $rm, $ru, $si, $du\n"); if (method=="REGISTER") { xlog("L_NOTICE", "*** register request from $si\n"); sl_send_reply("503", "No registrar service here"); return; } if (method=="INVITE" && ! uri=~"^sip:\+[1-9][0-9]+@") { xlog("L_NOTICE", "*** illegal sip uri: $ru\n"); sl_send_reply("484", "International number format required"); return; } # --- customer specific part --------------------------------------- undivert(2) # ------------------------------------------------------------------ if (method=="INVITE") { xlog("L_NOTICE", "*** reject invite from $si\n"); t_reply("403", "Forbidden"); return; } xlog("L_NOTICE", "*** $rm from $si\n"); t_relay(); } failure_route[2] { xlog("L_NOTICE", "*** in failure_route 1: rc: $rc, rr: $rr, rs: $rs\n"); if (! ds_next_domain()) { t_reply("503", "Service unavailable: no more gws"); return; } else { append_branch(); t_on_failure("2"); t_relay(); return; } } divert(-1) define(`CUST_IP_PRINT', `src_ip=='$1) define(`CUST_IP_FOREACH', `dnl ifelse($#, 0, `',dnl $#, 1, `CUST_IP_PRINT($1)',dnl `CUST_IP_PRINT($1) || CUST_IP_FOREACH(shift($@))')') define(`CUST_BLOCK', ` divert(2) if ( CUST_IP_FOREACH($1) ) { xlog("L_NOTICE", "*** $rm from '$2`, $si\n"); append_hf("P-Hint: from '$2`\n"); append_hf("Remote-Party-ID: <sip:+492319721231@62.191.185.47>;party=calling;screen=yes;privacy=off\n"); if (method=="INVITE") { strip(1); prefix("+'$3$4$5`"); } ds_select_domain("'$3`", "4"); if (method=="INVITE") { t_on_failure("2"); } xlog("L_NOTICE", "----> before t_relay <----\n"); t_relay(); xlog("L_NOTICE", "----> after t_relay <----\n"); return; } divert ') divert

2 2

[Users] Problem with IAX and Openser.
by david.castro 16 Jan '06

16 Jan '06

Thank you Daniel. The ouput is this: U 2006/01/16 11:49:22.392147 10.2.11.35:5062 -> 10.2.11.35:5060 INVITE sip:204@10.2.11.35 SIP/2.0..Via: SIP/2.0/UDP 10.2.11.35:5062;branch=z9hG4bK5c440b0c;rport..From: "David" <sip:"D avid"<sip:205@1021150@10.2.11.35:5062>;tag=as4a705794..To: <sip:204@10.2.11.35>..Contact: <sip:"David"<sip:205@1021150@ 10.2.11.35:5062>..Call-ID: 2d64d7461241acaf11db6bb21cd4830a@10.2.11.35..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Max -Forwards: 70..Date: Mon, 16 Jan 2006 10:49:22 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. .Content-Type: application/sdp..Content-Length: 463....v=0..o=root 15086 15086 IN IP4 10.2.11.35..s=session..c=IN IP4 1 0.2.11.35..t=0 0..m=audio 17016 RTP/AVP 0 8 4 111 18 3 97 7 110 5 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rt pmap:4 G723/8000..a=rtpmap:111 G726-32/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:3 GSM/8000..a=rtpmap: 97 iLBC/8000..a=rtpmap:7 LPC/8000..a=rtpmap:110 speex/8000..a=rtpmap:5 DVI4/8000..a=rtpmap:101 telephone-event/8000..a= fmtp:101 0-16..a=silenceSupp:off - - - -.. U 2006/01/16 11:49:22.392610 10.2.11.35:5060 -> 10.2.11.35:5062 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 10.2.11.35:5062;branch=z9hG4bK5c440b0c;rport=5062..From: "David" <sip:"David"<sip: 205@1021150@10.2.11.35:5062>;tag=as4a705794..To: <sip:204@10.2.11.35>..Call-ID: 2d64d7461241acaf11db6bb21cd4830a(a)10.2.1 1.35..CSeq: 102 INVITE..Server: OpenSer (1.0.0-tls (i386/linux))..Content-Length: 0..Warning: 392 10.2.11.35:5060 "Nois y feedback tells: pid=15870 req_src_ip=10.2.11.35 req_src_port=5062 in_uri=sip:204@10.2.11.35 out_uri=sip:204@10.2.11. 35 via_cnt==1".... U 2006/01/16 11:49:22.393443 10.2.11.35:5060 -> 10.2.11.35:5062 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 10.2.11.35:5062;branch=z9hG4bK5c440b0c;rport=5062..From: "David" <sip:"David"<sip :205@1021150@10.2.11.35:5062>;tag=as4a705794..To: <sip:204@10.2.11.35>;tag=329cfeaa6ded039da25ff8cbb8668bd2.1607..Call- ID: 2d64d7461241acaf11db6bb21cd4830a@10.2.11.35..CSeq: 102 INVITE..Server: OpenSer (1.0.0-tls (i386/linux))..Content-Le ngth: 0..Warning: 392 10.2.11.35:5060 "Noisy feedback tells: pid=15870 req_src_ip=10.2.11.35 req_src_port=5062 in_uri= sip:204@10.2.11.35 out_uri=sip:204@10.2.11.64:5060;transport=udp via_cnt==1".... U 2006/01/16 11:49:22.393731 10.2.11.35:5060 -> 10.2.11.35:5062 SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL)..Via: SIP/2.0/UDP 10.2.11.35:5062;branch=z9hG4bK5c440b0c;r port=5062..From: "David" <sip:"David"<sip:205@1021150@10.2.11.35:5062>;tag=as4a705794..To: <sip:204@10.2.11.35>;tag=329 cfeaa6ded039da25ff8cbb8668bd2.1607..Call-ID: 2d64d7461241acaf11db6bb21cd4830a@10.2.11.35..CSeq: 102 INVITE..Server: Ope nSer (1.0.0-tls (i386/linux))..Content-Length: 0..Warning: 392 10.2.11.35:5060 "Noisy feedback tells: pid=15870 req_sr c_ip=10.2.11.35 req_src_port=5062 in_uri=sip:204@10.2.11.35 out_uri=sip:204@10.2.11.64:5060;transport=udp via_cnt==1".. .. U 2006/01/16 11:49:22.397810 10.2.11.35:5062 -> 10.2.11.35:5060 ACK sip:204@10.2.11.35 SIP/2.0..Via: SIP/2.0/UDP 10.2.11.35:5062;branch=z9hG4bK5c440b0c;rport..From: "David" <sip:"Davi d"<sip:205@1021150@10.2.11.35:5062>;tag=as4a705794..To: <sip:204@10.2.11.35>;tag=329cfeaa6ded039da25ff8cbb8668bd2.1607. .Contact: <sip:"David"<sip:205@1021150@10.2.11.35:5062>..Call-ID: 2d64d7461241acaf11db6bb21cd4830a@10.2.11.35..CSeq: 10 2 ACK..User-Agent: Asterisk PBX..Max-Forwards: 70..Content-Length: 0.... I saw the ouput and I think it's all right except the address: <sip:"David"<sip:205@1021150@10.2.11.35:5062> What do you think about all that?

1 0

[Users] Problem with IAX and Openser.
by david.castro 16 Jan '06

16 Jan '06

I used ngrep -qt port 5060 but I don't get ouput in the screen, so I used: ngrep -qt, and the ouput was: U 2006/01/16 10:56:26.736394 10.2.11.50:4569 -> 10.2.11.35:4569 ..................204.."David"<sip:205@1021150..David..en..internal..david...................0V. U 2006/01/16 10:56:26.737313 10.2.11.35:4569 -> 10.2.11.50:4569 ..................205130695..david T 2006/01/16 10:56:26.736424 10.2.11.50:22 -> 10.2.11.35:60601 [AP] ].........I2..h.&......>,B.....X]OA.Q...@...b..a~..X.a..8...g.....)....y&.~.....@..+..<&....ly...b.{..Q...f..-..#...... ..d..\h....[.....He..&.]........._[..,.HZ|g..P.....r...f0.P.l{.....@..z........Y.&[......'..v......|.1.m.P.d.;M........ ..H$..T..<~z..Ra.. T 2006/01/16 10:56:26.736447 10.2.11.50:22 -> 10.2.11.35:60601 [AP] ..IM..D..X.Uv;.."..d...a.....q.Z:.7.l......Z..@.>-........~_..K^ U 2006/01/16 10:56:26.736538 10.2.11.50:4569 -> 10.2.11.35:4569 ............. b07125c8393cbdd4c3ca7b621bb678bc U 2006/01/16 10:56:26.736748 10.2.11.35:4569 -> 10.2.11.50:4569 .................. U 2006/01/16 10:56:26.736950 10.2.11.50:4569 -> 10.2.11.35:4569 ............ T 2006/01/16 10:56:26.737002 10.2.11.50:22 -> 10.2.11.35:60601 [AP] A.W..=..>n....S..>.....hI.PX@)..I.KHR.f2_j..5.....S.. ...h.....SH.og...zC.@...../.j..U?.Yk..%.m..........&...... T 2006/01/16 10:56:26.737065 10.2.11.50:22 -> 10.2.11.35:60601 [AP] r......]..G.n.w..H..'...rk...c..|0.`.......7.J........g....b.]Z...G.:^.f....p.[..b42..D.....t... U 2006/01/16 10:56:26.740189 10.2.11.35:4569 -> 10.2.11.50:4569 ............ U 2006/01/16 10:56:26.740263 10.2.11.35:4569 -> 10.2.11.50:4569 ............ U 2006/01/16 10:56:26.740314 10.2.11.50:4569 -> 10.2.11.35:4569 ............ U 2006/01/16 10:56:26.740381 10.2.11.50:4569 -> 10.2.11.35:4569 ............ T 2006/01/16 10:56:26.740535 10.2.11.50:22 -> 10.2.11.35:60601 [AP] ....z.O.Q,?....o...t...[..^.....0..2/X..98..M...'-...lP..I.......0..<.'.W..vDs....g...Y..G..v8.. T 2006/01/16 10:56:26.740566 10.2.11.50:22 -> 10.2.11.35:60601 [AP] ._.z...4....K^.#..J...`..l.0.)e..y.G.7.t.Y.t-W.5........41...K/.#..@..R......2T..m...J.K.:].....E.~le.....gh<-u% U 2006/01/16 10:56:26.740594 10.2.11.35:4569 -> 10.2.11.50:4569 ............*.& U 2006/01/16 10:56:26.740847 10.2.11.50:4569 -> 10.2.11.35:4569 ............ T 2006/01/16 10:56:26.741537 10.2.11.50:22 -> 10.2.11.35:60601 [AP] =.z..hym..f.z..H...8,.Uh:.....)r......j.&).".....=.Z$U... 13.....N......:.cn+L....../..:Q.....qX.."..Q\....3Z..N..1..y= ....!V....FR...j..MjpE=....j..r..k.......M..0..9.F...).M AI.-.....g..K.}S....8.z...N....3Z3.gQ?...,...#....H..5.dQ....- ...Tv.9z.....c.;..x....uq..........Ga....C... .......H}W})d....7...k..'d....LE..Kq..D!%4.yZ..?.'.......nBN.23...L)>..6. !.,......B.^.....v_.\.O.ER.J..}...7]8.bBj.. I saw this and I guess that sip packet is wrong, but I have in extexsion.conf: exten = _2XX,2,Dial(SIP/${EXTEN}@ser,100) I thought I was making a sip request. 2 notes: I have openser and asterisk in the same computer, asterisk has sip port 5062 and openser 5060. I test openser and asterisk talk one to one.

2 1

[Users] dbtable
by jammers yann 16 Jan '06

16 Jan '06

Can i have the role of each table in openser database ? ___________________________________________________________________________ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com

2 1

[Users] OpenSER TLS with Windows Messenger
by Cagri Koksal 16 Jan '06

16 Jan '06

Hi all, I'm trying to test openset TLS feature with windows messenger. I setup openser on Debian Linux and modified the openser.cfg and openssl.cnfas required by the TLS faq and docs at openser.org. When I try to sign in using windows messenger I got the following message from OpenSer: "tls-accept: Error in SSL:" At the same time windows messenger gives the following error messenger; "There was a problem verifying the certificate from the server ....." Here is an excerpt from my openser.cfg - disable_tls = 0 listen = tls:192.168.0.128:5061 tls_verify = 0 tls_require_certificate = 0 tls_method = TLSv1 ( I also tried SSLv23 ) tls_certificate = "/usr/local/etc/openser/tls/sipsrv_rootCA/sipsrv01/cert.pem" tls_private_key = "/usr/local/etc/openser/tls/sipsrv_rootCA/sipsrv01/privkey.pem" tls_ca_list = "/usr/local/etc/openser/tls/sipsrv_rootCA/sipsrv01/calist.pem" and an excerpt from my openssl.cnf dir = "/usr/local/etc/openser/tls/sipsrv_rootCA/demoCA" certs = $dir/certs crl_dir = $dir/crl database = $dir/index.txt new_certs_dir = $dir/newcerts certificate = $dir/cacert.pem serial = $dir/serial crl = $dir/crl.pem I also used the existing certificates that come with openser-tls package. It resulted the same. Any suggestion or guidance is appreciated, thanks. -Cagri Koksal

2 1

[Users] Radius Accounting
by martin＠campus-merseburg.de 16 Jan '06

16 Jan '06

Hi, why does OpenSER sends three Start-Radius-Packets on an Invite und and two on a bye? My config has these entries: [...] modparam("acc", "radius_flag", 3) [...] if (method=="BYE" || method=="CANCEL") { setflag(3); }; [...] } else if (method=="INVITE") { setflag(3); [...]

2 1

[Users] TM functions : merging
by Bogdan-Andrei Iancu 15 Jan '06

15 Jan '06

Hi everybody, I just committed on CVS head some changes into TM script functions - related functions are merged into new ones; the goal is to simplify the usage of these functions and to open way for NAPTR lookup. Following functions were merged: - t_relay_to_udp(), t_relay_to_tcp(), t_relay_to_tls() merged into t_relay(proto:host:port); - t_forward_nonack_uri(), t_forward_nonack_udp(), t_forward_nonack_tcp(), t_forward_nonack_tls() merged into t_forward_nonack(proto:host:port) and t_forward_nonack(); - t_replicate_udp(), t_replicate_tcp(), t_replicate_tls() merged into t_replicate(proto:host:port) Note that the functionality is not affected. To make the transition easier, the replaced functions are not removed yet, but marked as obsolete (via documentation and messages at runtime). The plan is to have them removed soon - a second set of changes into the API is to come. regards, bogdan

1 0

[Users] route on reply reason in failure_route
by Walter Schober 15 Jan '06

15 Jan '06

Can anyone please give me a hint how to access the reply reason in failure_route - or is it really not possible to trigger some event on the response _reason_. 0(704) SIP Reply (status): 0(704) version: <SIP/2.0> 0(704) status: <404> 0(704) reason: <CLIR Inactive> <----- I want this! There is a pseudo variable for that, but no function can search the replied reason! Is it really true, that you can use the reply status only (t_check_status("404"))? Using avpops the pseude variables $rr and $rs are <null> at that moment. Do I miss something? I'm doing that: failure_route[3] { if (t_check_status("404") ) { avp_printf("$reason", "The reply: $rr, $rs"); log("\n----------------------------------------\n"); avp_print(); log("\n----------------------------------------\n"); } } resulting in: 0(740) DEBUG:avpops:print_avp: p=0xb6177838, flags=2 0(740) DEBUG: id=<32> 0(740) DEBUG: val_str=<The reply: <null>, <null>> Any hints welcome. /Walter

2 4

[Users] Problem with IAX and Openser.
by david.castro 13 Jan '06

13 Jan '06

Hello. I am in a strange situation. I have two asterisk. Asterisk "A" makes a call for asterisk "B" by IAX. Asterisk "B" recives the call and delivers it to Openser by SIP. The problem is openser printing this in the screen: ERROR: parse_to : unexpected char ["] in status 5: <<"David" <sip:>> . ERROR:parse_from_header: bad from header ERROR: new_t: no valid From in INVITE ERROR: t_newtran: new_t failed ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) Do you know the reason? Is this a bug? Which is guilty, asterisk or openser? I have this problem only in this scenario. Thank you. -- -------------------------------- David Castro Mosquera Adianta Tel: (34) 902 999 178 Fax: (34) 902 501 437 Email: david.castro(a)adianta.net Web: www.adianta.net --------------------------------

2 1

[Users] New functionality : Direction detection in ACC module
by Bogdan-Andrei Iancu 13 Jan '06

13 Jan '06

Hi everybody, I just committed on CVS head a new functionality that was requested by Lenir <lenirsantiago(a)yahoo.com> some time ago. It's about detecting and compensating in the ACC records of the direction flow for sequential requests. For more details about it and about how to used it, please see the online documentation: http://openser.org/docs/modules/1.1.x/acc.html#AEN625 Any testing/feedback is highly appreciated ;). regards, bogdan

1 0

[Users] delete specific contacts from location
by Mike Williams 13 Jan '06

13 Jan '06

Is it possible to delete a specific contact location for a user, and by using openserctl? I see that adding and deleting a certain contact is on the openser roadmap, and if this means that this functionality has made it into openser but not into openserctl, than I'm willing to try to add it myself. Any advice where to start is appreciated. Mike Williams

2 1

[Users] Functionality
by Andréa Giordanna 13 Jan '06

13 Jan '06

Hi all I need a functionality on openser. The functionality would allow users to save on the server - a message, - a time and date - and a group of uris, and at the given time, the server would send the message to each one of the group of uris. This functionality would be useful for users to send reminders of events to a list of uris. Are there any modules on openser that do something like that? If not, how could I implement it? Thanks Andrea Giordanna --------------------------------- Yahoo! doce lar. Faça do Yahoo! sua homepage.

2 3

[Users] about sems
by unplug 12 Jan '06

12 Jan '06

Since I have no idea about asterisk, and I would like to ask can I use sems instead of asterisk to provide same function. Any openser+sems howto available for fedora core4?

3 2

[Users] Openser as a Outbound Proxy
by david.castro 12 Jan '06

12 Jan '06

This is the sip.conf from asterisk: ########################################################################### [general] bindport=5062 bindaddr=0.0.0.0 ;canreinvite=no disallow=all ;allow=all register => asterisk:asterisk@my_openser_ip ;you can found more info of this in http://www.voip-info.org/wiki/ [ser] allow=ulaw allow=alaw allow=g723.1 allow=g726 allow=g729 allow=gsm allow=ilbc allow=lpc10 allow=speex allow=adpcm type=friend ; We allow incoming and outgoing calls. Use peer if you are only doing MWI context=internal ; This is the context incoming calls land in host=ip_ser ; This is the hostname or IP address of your SER server ;fromdomain=ser.server.rld ; This is your SER_DOMAIN insecure=very ; This allows incoming calls from the phones routing through ser to be passed into asterisk dtmfmode=rfc2833 This is the openser.cfg ###################################################################### # # $Id: openser.cfg,v 1.5 2005/10/28 19:45:33 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (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/openser_fifo" # # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls #tls_verify = 1 #tls_require_certificate = 0 #tls_require_certificate = 1 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_certificate = "/usr/local/etc/openser/tls/openser1/cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_private_key = "/usr/local/etc/openser/tls/openser1/privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" #tls_ca_list = "/usr/local/etc/openser/tls/openser1/calist.pem" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database 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/textops.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/uac.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" #loadmodule "/usr/local/lib/openser/modules/avpops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/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{ xlog("yo soy <$si> y el destino es <$tu> <$td> <$du> <$rm> \n"); # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # ----------------------------------------------------------------- # Handle NOTIFY requests from Sipura's to keep-alive NAT # ----------------------------------------------------------------- if ((is_method("NOTIFY")) && (! uri=~"sip:.*[@]+.*")) { xlog("entre en notify \n"); sl_send_reply("200", "OK keep-alive"); return; } # ----------------------------------------------------------------- # Only allow known sip messages,ack, pq no hace falta responder. # ----------------------------------------------------------------- if (!is_method("INVITE|REGISTER|BYE|ACK|CANCEL")) { xlog("Estoy en metodo no diponible y que es <$rm>\n"); sl_send_reply("501", "Not implemented here"); return; }; # 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()) { xlog("entree en loose route\n"); # mark routing logic in request append_hf("P-hint: rr-enforced\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 (uri=~"^sip:*.") { if (method=="ACK" && (src_ip!=XXXXXX)){ # Uncomment this if you want to use digest authentication if (!www_authorize("XXXXXXX", "subscriber")) { www_challenge("XXXXXXXX", "0"); exit; }; consume_credentials(); xlog("Voy a mandar respuesta a un ACK, yo soy <$si> y el destino es <$tu> <$td> <$du>\n"); route(1); exit; }; if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("XXXXX", "subscriber")) { www_challenge("XXXXX", "1"); exit; }; xlog("Me llego un register y yo soy <$si> y el destino es <$tu> <$td> <$du>\n"); consume_credentials(); save("location"); exit; }; if (method=="INVITE") { # Uncomment this if you want to use digest authentication if (!www_authorize("XXXXX", "subscriber") && (src_ip!=XXXXXX) ) { www_challenge("XXXXX", "0"); exit; }; # consume_credentials(); xlog("Un tio mando un invite <$si> y el destino es <$tu> <$td> <$du> \n"); sl_send_reply("100", "Trying"); lookup("aliases"); #Si es de asterisk y no encuentra el telefono digo que no esta. if (!lookup("location") && (src_ip==XXXXXXXX) ) { #xlog("no encuentra el destino\n"); sl_send_reply("404", "Not Found"); exit; } else if (lookup("location") && (src_ip==XXXXX)){#if a call come from * and sip phone online i route to ring sl_send_reply("180", "Ringing"); route(2); return; } else {#otherway i forward to *. append_hf("P-hint: usrloc applied\r\n"); #sl_send_reply("180", "Ringing"); route(1); return; }; }; if (method=="CANCEL" && (src_ip==XXXXX)) {#Si viene de asterisk, lo mando con estado al telefo. #xlog("Me llego un cancel de asterisk\n"); route(2); return; } else if (method=="CANCEL" && (src_ip!=XXXXXX)) { #Si no viene de asterisk... # Uncomment this if you want to use digest authentication if (!www_authorize("XXXXXX", "subscriber")) { www_challenge("XXXXXX", "0"); exit; }; consume_credentials(); #xlog("Me llego un cancel de NO asterisk\n"); lookup("aliases"); if (!lookup("location")) {#primero compruebo que este numero existe. #xlog("no encuentra el destino\n"); sl_send_reply("404", "Not Found"); exit; } else { route(1); #Se lo mando a asterisk. return; }; }; if (method=="BYE") { # Uncomment this if you want to use digest authentication if (!www_authorize("XXXXX", "subscriber") && (src_ip!=XXXXXX) ) { www_challenge("XXXXX", "0"); exit; }; consume_credentials(); xlog("Me llego un BYE \n"); if (src_ip!=XXXXXXX){ route(1); } else { route(2); }; sl_send_reply("200", "OK"); exit; }; }; } #I route to asterisk here route[1] { xlog("Estoy en route(1) \n"); forward(XXXXX,5062); return; } #I route from asterisk to sip phones here route[2] { xlog("Estoy en route(2)\n"); if (!t_relay()) { #xlog("Dentro de error del t_relay del 2\n"); sl_reply_error(); }; return; } ################################################### Notes: XXXX=ip_asterisk This configuration works in the same net. Now, I'm trying to work behind a nat. This configuration don't works behind a nat. I don't challenge asterisk, if a call comes from asterisk_ip, I trust. Bye.

1 0

[Users] NOTIFY and RFC
by Helge Waastad 12 Jan '06

12 Jan '06

Hi, My question today is more regarding RFC's. I've tried to find out if NOTIFY and Event: message-summary needs a Contact header. At least my Cisco gateway discards Notify's without a Contact header (400 Bad Request - 'Malformed/Missing Contact field') Is this a Cisco bug or SHOULD/MUST a NOTIFY have Contact of the sender? br hw

2 2

[Users] Re: Users Digest, Vol 8, Issue 23
by unplug 12 Jan '06

12 Jan '06

As I know, each table is assigned to a module. For example, avpops module will use table usr_preferences. The usage of this module can be found in the module document. > Date: Wed, 11 Jan 2006 16:20:58 +0100 (CET) > From: jammers yann <foudelinux(a)yahoo.fr> > Subject: [Users] dbtable > To: users(a)openser.org > Message-ID: <20060111152058.85120.qmail(a)web25806.mail.ukl.yahoo.com> > Content-Type: text/plain; charset=iso-8859-1 > > Can i have the role of each table in openser database ? >

1 0

[Users] web interface
by jammers yann 11 Jan '06

11 Jan '06

hi! do we have a web interface for openser like serweb for ser ? Or can we easily integarte serweb with openser thanks ___________________________________________________________________________ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com

4 3

[Users] Re: OpenSER and transport selection (SRV and NAPTR)
by Greg Fausak 11 Jan '06

11 Jan '06

Howdy guys, I completely missed this thread. How is this going? I just read the roadmap which indicates NAPTR lookup is on the list. I've been maintaining NAPTR and SRV records looking forward to the day when t_relay() forwards using NAPTR (and SRV). Does that mean that the t_relay() will maintain the state, slogging through the NAPTR/SRV records appropriately, without me having to do additional logic??? Specifically, I'm keen on the SRV serial forking. -g Hi Bogdan, > I meant t_reply() will keep its behaviour as looking into URI for the > destination - but it will incorporate the NAPTR lookup. Great - this answers my question. > to response also to on earlier question, regarding the timing > - I say 3 months are more than enough ;). Excellent. Many thanks for your help (and patience ;) --Joachim > Joachim Fabini wrote: > > >Hi Bogdan, > > > > > > > >>indeed, there was a misunderstanding :) .t_relay() with no > >>param will be kept with the current functionality. > >>Or you suggest to be able to specify only a different proto > >>or port from the RURI? > >> > >> > > > >I did not yet find the primitive which is supposed to > >statefully relay and do the following: > >1. NAPTR query on the transport to use PLUS > >2. _exactly_ what t_relay() does now for the > > transport returned by the NAPTR query. > > > >You say that t_relay() is kept with the current functionality. > >Does this mean no NAPTR, or will t_relay() be extended to use > >NAPTR before SRV/A query? If the latter then everything is OK. > > > >Otherwise we have the alternatives: > > > >t_relay() -> Stateful relaying according to destination > > address using the incoming transport (no NAPTR). > >t_relay_to() -> Stateful relaying to a specific host (you said > > that <host> is mandatory here) using NAPTR to > > determine the transport. > > > > > >To summarize: > >My point is that none of these two primitives covers the case > >when the message is to be relayed to the next hop based only > >on the destination address _and_ using the transport selected > >by the destination proxy (determined via NAPTR query). > > > >Except if either a) t_relay() does NAPTR or b) the host > >parameter in t_relay_to() is optional. > > > >--Joachim > > > > > > > > > -- Greg Fausak greg(a)thursday.com

2 2

[Users] Re: TLS problem
by david.castro 11 Jan '06

11 Jan '06

Thank you, I pleased with you if you could explain how I can implement that, or where I can found information about it. Thanks.

2 1

[Users] Openser as a Outbound Proxy
by david.castro 11 Jan '06

11 Jan '06

I don't know if this help you, but I try. I have a similar scenario. I have openser, which deals with sip registers. If a sip phone make a call, I send the request to asterisk. Asterisk dispatch to PSTN or go back to openser (if this call is for sip). Asterisk deal the dialplan and cdr-csv and it's a gateway to PSTN. This is an idea. If you want i can post the config files. Bye. -- -------------------------------- David Castro Mosquera Adianta Tel: (34) 902 999 178 Fax: (34) 902 501 437 Email: david.castro(a)adianta.net Web: www.adianta.net --------------------------------

2 1

Re: Re: [Users] Help: how to use pa module
by Kenny Yeh 11 Jan '06

11 Jan '06

samuel,hi yeah,as you said I changed cfg file,but ser is still cann't be started. Did you try? ======= 2006-01-11 16:43:02 您在来信中写道：======= >First of all: pa module is not stable and might not work. > >The other thing is that handle_subscription only accepts registrar or >jabber as parameters, so you have to change debian for registrar: > > handle_subscription("registrar"); > > >Luck!!! >Samuel. > >2006/1/11, Kenny Yeh <kenny(a)artdio.com.tw>: >> Hi,all >> >> I complied pa module in the right directory,but as the help file I cann't start openser right.my cfg file is below: >> >> #I added pa.so file >> loadmodule "/usr/local/lib/openser/modules/pa.so" >> # >> modparam("pa", "default_expires", 3600) >> handle_subscription("debian") >> >> debian is my main domain >> >> what's wrong with it? I need a help !!!! >> >> >> Kenny yeh >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> = = = = = = = = = = = = = = = = = = = = 　　　　　　　　致礼！　　　　　　　　Kenny Yeh 　　　　　　　　kenny(a)artdio.com.tw 　　　　　　　　　　2006-01-11

2 1

[Users] Help: how to use pa module
by Kenny Yeh 11 Jan '06

11 Jan '06

Hi,all I complied pa module in the right directory,but as the help file I cann't start openser right.my cfg file is below: #I added pa.so file loadmodule "/usr/local/lib/openser/modules/pa.so" # modparam("pa", "default_expires", 3600) handle_subscription("debian") 　 debian is my main domain what's wrong with it? I need a help !!!! Kenny yeh

2 1

[Users] Help: presence funtion
by Kenny Yeh 11 Jan '06

11 Jan '06

Hi,all I have a question: OpenSER can be a presence server? how ? I try to test by msn5.1,but OpenSER cann't recive 'subscribe' request. Can I load pa module to help that? BR! Kenny

1 0

[Users] Via branch parameter
by Scott MacGregor 10 Jan '06

10 Jan '06

Is there a way to change the branch parameter in any of the Via headers in an INVITE? I am getting a 482 loop detected which I think is due to the same branch parameter in two requests. Thanks, Scott __________________________________________ Yahoo! DSL Something to write home about. Just $16.99/mo. or less. dsl.yahoo.com

3 2

[Users] tracking down problems adding permanent usrloc record
by Mike Williams 10 Jan '06

10 Jan '06

On Monday 02 January 2006 11:11, Juha Heinanen wrote: > Mike Williams writes: > > I created the account 11112223333(a)mydomain.net to test with, and then I > > attempted to use: > > > > openserctl ul add 11112223333(a)etcvoip.net sip:18122228203@etcvoip.net > > > > but it returned: > > > > error: 400; check if you use aliases in OpenSER > > > > I added the alias_db module to my config, but I still get errors. Any > > advice? Should I add them manually to the location database? > > i have never used openserctl. my management application writes > permanent contacts directly to location table. remember to tell via > flags field that registration is permanent. > > -- juha I modified the openserctl script to give more output: 400 table (aliases) not found error: 400; check if you use aliases in OpenSER on `openserctl ul add 11112223333(a)etcvoip.net sip:18122228203@etcvoip.net` I traced this message back to the ul_show_contact function in ul_fifo.c. Everything is fine until it gets to line 573, where it calls fifo_find_domain, which sets d=0. Then, in line 575, the error message given above is returned because d if now false. static inline void fifo_find_domain(str* _name, udomain_t** _d) { dlist_t* ptr; ptr = root; while(ptr) { if ((ptr->name.len == _name->len) && !memcmp(ptr->name.s, _name->s, _name->len)) { break; } ptr = ptr->next; } if (ptr) { *_d = ptr->d; } else { *_d = 0; } } I'm not a C programmer, so I don't really understand what's happening here, but it appears to be looping through a list to find a domain, checking to be sure their lengths and values match. It keeps looping until it finds one, and then quits on the first one. It appears that ptr doesn't point to anything in my case; it fails the next if, setting the passed-by-reference d to 0, leading me to think that it hasn't found any domains, or couldn't get a domain list in the beginning. What does this "ptr = root;" mean? I then tried running `openserctl domain show`, only to get: 500 command 'domain_dump' not available Now, I checked the actual database openser.domain table, and my domain appeared correctly. I am also loading the domain module on openser startup. Any ideas? I realize this is getting awfully deep into code, but I hope someone can help. It's probably a simple error on my part. Thanks in advance. Mike Williams

4 4

[Users] LCR module
by Ovidiu Sas 10 Jan '06

10 Jan '06

Hi all, The new lcr module has now two modes of operation controlled by the direct_db parameter: - direct_db = 1 (no DB caching) This is the old behavior of the lcr module - direct_db = 0 (DB caching - default one) In this mode, the 1.7 TODO issue from the README file is addressed: no more SQL query in load_gws() function. In addition to this, the From URI from the lcr table is now a POSIX regular expression, with the following extension: if the From URI is an empty string, it will be considered as a universal match - ie .* without invoking the regexec function. Migrating from the old mode to the new mode requires the from_uri column from the lcr table to be updated to use POSIX regular expressions instead of the special _ and % characters. Example: old from_uri -> new from_uri ------------------------------ % -> 123% -> 123.* regards Ovidiu Sas

1 0

[Users] TLS problem
by David 10 Jan '06

10 Jan '06

Hello. I'm trying to use TLS with openser. I've made the steps from the tutorial:http://openser.org/docs/tls.html. I have "tls_verify = 1" and "tls_require_certificate = 1". I have a phone without TLS support, but it registers in openser. I think, this shouldn't happend. Anybody has ideas? Second question: I used netstat to verify if somebody is listening on 5060 and 5061, but if I put ip_openser:5061 in the phone, I have an icmp message with "destination unreachable". What's happening? Thanks

3 4

[Users] LCR problem
by Pepe 10 Jan '06

10 Jan '06

Hello, Im making tests and its not a LCR problem, its a problem from my GW2, when I use it for first option, it fails too, here you have the ngrep, ClientA --> Proxy --> GW2 (192.168.10.93) (192.168.10.91) (195.219.74.166) Regards The problem is that the BYE request will be handled by your LCR logic. The BYE request should be route in the loose_route block as it is an in-dialog request. Maybe the BYE sent from the gateway is not correct. Please post a ngrep dump (ngrep -t -W byline port 5060) regards klaus Pepe wrote: > Hello, > > Im configuring Openser with LCR module and Im having an extrange > behavior, I have 2 gateways, GW1(preference1) and GW2(preference2), > > GW1(pref.1) > / \ > ClientA --> OpenSer --> Client B > \ GW2 (pref.2) / > > > When I call from Client A to Client B using GW1, all works fine, its the > same when hang up Client B or Client A, but when GW1 fail(I provoke it > changing codec) and use failure route (GW 2) then if Client A hang up > all works fine, but the problem is when is Client B who hang up, its > like a new conversation, GW 2 send BYE to openser and Openser just send > "503 Service Not avilable - No gateways" to GW2, but doesnt send nothing > to ClientA, any idea ???? > > > Thx in advance >

2 6

[Users] direct manipulation of DB
by Norman Brandinger 09 Jan '06

09 Jan '06

For example, using PHP, I have found that the code below works well: $a1="$subscriber:$domain:$password"; $ha1=md5($a1); $a1b="$subscriber@$domain:$domain:$password"; $ha1b=md5($a1b); Regards, Norm

2 1

[Users] problem accounting of ACKs
by Klaus Darilion 09 Jan '06

09 Jan '06

Hi! I have problems accounting ACK (e2e ACKs after INVITE-200 Ok). I have the following setup: client1 proxy1 proxy2 proxy3 client2 |---INVITE--->| | | | | rewritehostport | | | | t_relay | | | | |--INVITE->| | | | | enum_lookup | | | | t_relay | | | | |-INVITE-->| | | | | lookup() | | | | t_relay | | | | |-----INVITE->| | | | | | | | | |<----200 OK--| | | |<--200 OK-| | | |<--200 OK-| | | | <--200 OK---| | | | | | | | | | ---ACK----> | | | | | loose_route | | | | |----ACK-> | | | | | loose_route | | | | |---ACK--->| | | | | loose_route | | | | |---ACK------>| | | | | | All proxies do accounting. I do see the INVITE and BYE requests accountend by all proxies, but the ACK is only accounted by proxy1. All proxies are configured nearly identical: # accounting parameters modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 0) modparam("acc", "multi_leg_enabled", 0) modparam("acc", "failed_transaction_flag", 4) # radius, syslog and DB modparam("acc", "log_flag", 1) # syslog modparam("acc", "db_flag", 2) # DB modparam("acc", "db_missed_flag", 3) # DB missed calles table modparam("acc", "swap_direction", 1) and all the flags are set for sure: route{ setflag(1);setflag(2); setflag(3);setflag(4); Does somebody had a similar problem? Are there any known issues how ACKs are handled? Studying the debug log. it looks like proxy 1 identifies the ACK as belonging to a transaction, whereas proxy 2 does not find the transaction (full debug at the end of the mail): proxy 1: ----------- t_lookup_request: start searching: hash=6242, isACK=1 parse_headers: flags=38 DEBUG: t_lookup_request: e2e proxy ACK found parse_headers: flags=8 DEBUG: totag for e2e ACK found: 0 DBG: trans=0x406b1940, callback type 4, id 0 entered parse_headers: flags=258 ACC: request acknowledged: method=ACK, i-uri=sip:user201@..... proxy 2: ----------- t_lookup_request: start searching: hash=6242, isACK=1 DEBUG: proceeding to pre-RFC3261 transaction matching DEBUG: t_lookup_request: no transaction found DEBUG:tm:t_relay: forwarding ACK statelessly DEBUG: mk_proxy: doing DNS lookup... parse_headers: flags=1000 check_via_address(10.10.0.41, 10.10.0.41, 0) Sending: ACK sip:user201@10.10.1.50:8737 SIP/2.0^M To: <sip:+43201@proxy1 Thus, why is there a matching transaction in proxy 1, but not in proxy 2? Why are ACKs not accounting in every case? Thanks Klaus proxy 1 (full debug): ------------------- SIP Request: method: <ACK> uri: <sip:user201@10.10.1.50:8737> version: <SIP/2.0> parse_headers: flags=2 DEBUG: add_param: tag=2b4e3b2d DEBUG:parse_to:end of header reached, state=29 DEBUG: get_hdr_field: <To> [52]; uri=[sip:+43201@proxy1.ienum.labs.nic.at] DEBUG: to body [<sip:+43201@proxy1.ienum.labs.nic.at>] Found param type 232, <branch> = <z9hG4bK-d87543-685643142-1--d87543->; state=6 Found param type 235, <rport> = <n/a>; state=17 end of header reached, state=5 parse_headers: Via found, flags=2 parse_headers: this is the first via After parse_msg... preparing to run routing scripts... DEBUG : sl_filter_ACK: to late to be a local ACK! DEBUG: add_param: tag=f937ab7a DEBUG:parse_to:end of header reached, state=29 xl_printf: final buffer length 186 parse_headers: flags=100 get_hdr_field: cseq <CSeq>: <2> <ACK> DEBUG:maxfwd:is_maxfwd_present: value = 70 parse_headers: flags=200 is_preloaded: No grep_sock_info - checking if host==us: 10==10 && [10.10.1.50] == [10.10.0.41] grep_sock_info - checking if port 5060 matches port 8737 grep_sock_info - checking if host==us: 10==10 && [10.10.1.50] == [10.10.0.41] grep_sock_info - checking if port 5060 matches port 8737 grep_sock_info - checking if host==us: 10==10 && [10.10.1.50] == [10.10.0.41] grep_sock_info - checking if port 5061 matches port 8737 check_self: host != me grep_sock_info - checking if host==us: 10==10 && [10.10.0.41] == [10.10.0.41] grep_sock_info - checking if port 5060 matches port 5060 after_loose: Topmost route URI: 'sip:10.10.0.41;lr;ftag=f937ab7a' is me after_loose: URI to be processed: 'sip:10.10.0.41:5062;r2=on;ftag=f937ab7a;lr=on' after_loose: Next URI is a loose router parse_headers: flags=ffffffffffffffff DEBUG: get_hdr_body : content_length=0 found end of header DEBUG: t_newtran: msg id=2 , global msg id=0 , T on entrance=0xffffffff parse_headers: flags=ffffffffffffffff parse_headers: flags=78 t_lookup_request: start searching: hash=6242, isACK=1 parse_headers: flags=38 DEBUG: t_lookup_request: e2e proxy ACK found parse_headers: flags=8 DEBUG: totag for e2e ACK found: 0 DBG: trans=0x406b1940, callback type 4, id 0 entered parse_headers: flags=258 ACC: request acknowledged: method=ACK, i-uri=sip:user201@10.10.1.50:8737, o-uri=sip:user201@10.10.1.50:8737, call_id=9c5e0930eb2b8e5a, from="user101"<sip:user101@proxy1.ienum.labs.nic.at>;tag=f937ab7a, code=200 parse_headers: flags=258 grep_sock_info - checking if host==us: 10==10 && [10.10.1.50] == [10.10.0.41] grep_sock_info - checking if port 5060 matches port 8737 grep_sock_info - checking if host==us: 10==10 && [10.10.1.50] == [10.10.0.41] grep_sock_info - checking if port 5060 matches port 8737 grep_sock_info - checking if host==us: 10==10 && [10.10.1.50] == [10.10.0.41] grep_sock_info - checking if port 5061 matches port 8737 check_self: host != me Route ftag and From tag are same DEBUG:tm:t_relay: forwarding ACK statelessly DEBUG: mk_proxy: doing DNS lookup... check_via_address(10.10.1.51, 10.10.1.51, 0) Sending: ACK sip:user201@10.10.1.50:8737 SIP/2.0^M To: <sip:+43201@proxy1 proxy 2 (full debug): ------------------- SIP Request: method: <ACK> uri: <sip:user201@10.10.1.50:8737> version: <SIP/2.0> parse_headers: flags=2 DEBUG: add_param: tag=2b4e3b2d DEBUG:parse_to:end of header reached, state=29 DEBUG: get_hdr_field: <To> [52]; uri=[sip:+43201@proxy1.ienum.labs.nic.at] DEBUG: to body [<sip:+43201@proxy1.ienum.labs.nic.at>] Found param type 232, <branch> = <0>; state=16 end of header reached, state=5 parse_headers: Via found, flags=2 parse_headers: this is the first via After parse_msg... preparing to run routing scripts... DEBUG : sl_filter_ACK: to late to be a local ACK! DEBUG: add_param: tag=f937ab7a DEBUG:parse_to:end of header reached, state=29 xl_printf: final buffer length 186 parse_headers: flags=100 Found param type 232, <branch> = <z9hG4bK-d87543-685643142-1--d87543->; state=6 Found param type 235, <rport> = <6477>; state=16 end of header reached, state=5 parse_headers: Via found, flags=100 parse_headers: this is the second via get_hdr_field: cseq <CSeq>: <2> <ACK> DEBUG:maxfwd:is_maxfwd_present: value = 69 parse_headers: flags=200 is_preloaded: No grep_sock_info - checking if host==us: 10==10 && [10.10.1.50] == [10.10.0.41] grep_sock_info - checking if port 5062 matches port 8737 grep_sock_info - checking if host==us: 10==10 && [10.10.1.50] == [10.10.0.41] grep_sock_info - checking if port 5062 matches port 8737 grep_sock_info - checking if host==us: 10==10 && [10.10.1.50] == [10.10.0.41] grep_sock_info - checking if port 5063 matches port 8737 check_self: host != me grep_sock_info - checking if host==us: 10==10 && [10.10.0.41] == [10.10.0.41] grep_sock_info - checking if port 5062 matches port 5062 after_loose: Topmost route URI: 'sip:10.10.0.41:5062;r2=on;ftag=f937ab7a;lr=on' is me grep_sock_info - checking if host==us: 10==10 && [10.10.0.41] == [10.10.0.41] grep_sock_info - checking if port 5063 matches port 5063 after_loose: URI to be processed: 'sip:10.10.0.42:5063;transport=tls;r2=on;ftag=f937ab7a;lr=on' after_loose: Next URI is a loose router DEBUG: has_totag: totag found parse_headers: flags=ffffffffffffffff DEBUG: get_hdr_body : content_length=0 found end of header DEBUG: t_newtran: msg id=1 , global msg id=0 , T on entrance=0xffffffff parse_headers: flags=ffffffffffffffff parse_headers: flags=78 t_lookup_request: start searching: hash=6242, isACK=1 DEBUG: proceeding to pre-RFC3261 transaction matching DEBUG: t_lookup_request: no transaction found DEBUG:tm:t_relay: forwarding ACK statelessly DEBUG: mk_proxy: doing DNS lookup... parse_headers: flags=1000 check_via_address(10.10.0.41, 10.10.0.41, 0) Sending: ACK sip:user201@10.10.1.50:8737 SIP/2.0^M To: <sip:+43201@proxy1

2 1

[Users] what's about srtp on openser?
by Kenny Yeh 09 Jan '06

09 Jan '06

Bogdan-Andrei Iancu,hello Ask for you again. :) I am succesful on openser for TLS connected by SIPp,but not minisip :( thanks for your more help! Now I step to srtp! Can I use SRTP to build peer to peer communication from openser? if outbound via openser.or do you learn more about srtp of SIP. Best Regard! Kenny from China Shanghai 2006-1-5

3 2

Re: Re: [Users] help for TLS on openser
by Kenny Yeh 09 Jan '06

09 Jan '06

Bogdan-Andrei Iancu,hello Thanks for your help!!!! I try 'netstat -ual' but nothing is list; I try 'netstat -al' list is below: Active Internet connections (servers and established) Proto Recv-Q Send-Q Local Address Foreign Address State tcp 0 0 debian1:sip-tls *:* LISTEN tcp 0 0 localhost:mysql *:* LISTEN tcp 0 0 *:ftp *:* LISTEN tcp 0 0 localhost:smtp *:* LISTEN tcp6 0 0 *:ssh *:* LISTEN We can see 'debian1' :sip-tls,but I cann't see service port 5061 ,why? or must I connect to server on 5061 port,unlees I cann't see it? Thanks you again. ======= 2006-01-05 02:50:15 您在来信中写道：======= >Hi Kenny, > >Kenny Yeh wrote: > >>users，hello >> >> >> Happly new year! >> >> My problem: >> >> 1.I start openser with TLS support,follow the TLS help file,create a ca/demoCA/ and create cert.pem,private.pem,key.pem.openser is start well,but I cann't list the listen ports by openserctl ps,how I can be sure openser is started by TLS support? >> >> >use "netstat -ual" to check for port 5061 (TLS one). > >> 2.My client is minisip phone that TLS support.but when I start minisip to registar,the error messenge is "exception caught when creating TLS server",I donn't know what's happend. >> >> >ask the minisip guys or their lists. > >regards, >bogdan > >> Please help me ! >> >> >> >>　　　　　　　　致 >>礼！ >> >> >>　　　　　　　　Kenny Yeh >> 上海金叶通讯科技有限公司 >> TEL:8621-6421-6758 ext.311 >>　　　　　　　　kenny(a)artdio.com.tw >>　　　　　　　　　　2006-01-04 >> >> >>------------------------------------------------------------------------ >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> > = = = = = = = = = = = = = = = = = = = = 　　　　　　　　致礼！　　　　　　　　Kenny Yeh 　　　　　　　　kenny(a)artdio.com.tw 　　　　　　　　　　2006-01-05

2 1

[Users] Re:Functionality
by ye jian 09 Jan '06

09 Jan '06

Hi, I think what's you said is a p2p on SIP.that's a good idea for SIP,but who can make it to reality? I cann't ,who? -- 美好的一天,由你和我来创造!

1 0

Re: [Users] snom registration expiry
by Christian Benke 09 Jan '06

09 Jan '06

Hello Klaus! Thanks for your reply, you have triggered some new ideas. of course i should have thought about checking back if any packets are received by the openser host after the line is dead but i thought if it still works with ser, it should work with openser too(and i still have no problems with a second line registered at ser@port5060 while the registration at openser@5061 is lost). ok, the result of the ngrep was that no packets appear on the openser-host when i try to call from the (dead) line. i've made some tracing both on the snom and the ser/openser-host as you suggested. To my surprise, the snom seems to be the device that doesn't behave as expected - it doesn't re-register on the openser-configuration! I have made the attached traces in this way: First i've set the account-details on the snom with 103456(a)sip.de.domain.cc,then rebooted the phone and started the trace on the siprouter-host on port 5060(ser), then changed the configuration to 112233(a)siptest1.domain.cc, rebooted the phone and made the second trace on port 5061(openser). In both cases i didn't touch the phone or make any calls after the reboot to get a pure registration-trace. When i use the account 112233(a)siptest1.domain.cc, the snom registers once and doesn't try to re-register anymore. With the account 103456(a)sip.de.domain.cc, the snom re-registers every 5 minutes, not the expected 10 minutes, but at least it does re-register. When switching the account-details, i didn't do anything but changing the login-details(username, password, registrar-url), the rest of the configuration has not been touched. The phone-firmware is 3.60w. There doesn't seem to be a difference between the siplogs, at least i can't see it. i'm really puzzled right now... 8-| I've attached the siptrace from the ser/openser-host: SER_host_siptrace.txt/OPENSER_host_siptrace.txt the syslog from snom190: SER_snom190_log.txt/OPENSER_snom190_log.txt and the siptrace from snom190: SER_snom_siptrace.txt/OPENSER_snom190_siptrace.txt I have tried this several times now, changing the configuration and rebooting, same results everytime. on SER it's reregistering exactly every 5 minutes, on OPENSER it isn't. I hope all this information is not too much and you have the time to sniff through. thanks! christian p.s.:i've also thought about a error in the srv-records on my nameservers, but they look exactly the same for all sub-domains, the only difference is the port change from 5060 to 5061... On Thu, 05 Jan 2006 10:58:46 +0100 Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: > Hi Christian! > The more interesting part would be the SIP signaling between the SNOM and the proxy. If registration is lost, that means that: > - SNOM stops sending REGISTER or > - SNOM still sends REGISTER but they are not accepted by openser Please set exires back to 10min and watch the SIP signaling: > - using "ngrep -t -W byline port 5060" on the proxy > - using the logging feature of the snom phone > Please post this logs. > regards > klaus > Christian Benke wrote: > > Hello! > > > > I've been running my experimental(production in the near future) environment with ser and a mysql-db for some months now but want to switch > > to postgres and openser soon. i'm using a very simple routing as i didn't > > have the time to dig deeper into the ser-routing-language. > > > > i've already adapted my current config-file to openser and the routing works as expected. still, there is one major problem i could not cope with yet: Expiry. > > > > i'm using a snom 190 and had the "proposed expiry" > > set to 10min with ser, without any registration-problems or > > registration-loss. > > with openser with the same settings on the snom i loose my registration > > within a few minutes after phone-reboot, when setting "proposed expiry" to > > 1min on the snom it works without problems endlessly - but setting this > > generally to 1min on all the phones that will register at openser doesn't > > seem to be a good idea and i guess there are other ways. > > > > I've tried to force it with the modparams min_expire 100, max_expire 1800 > > and default_expire 600 > > but the result is still the same, calls are possible after rebooting the > > phone, but after some minutes(below 10min), i loose registration. I had > > similar problems due to a firewall closing the connection, but that occured when i had set a 60min expiry on the snom(with ser), the 10min timeout works fine with the same snom 190 and ser. > > I've searched a lot for > > expiry-problems but only found the modparams above that could be responsible for this. > > Sorry if my explanations sound simple and untechnical, > > but i'm pretty unexperienced with ser/openser yet so i try to explain it > > the way i understand it, simple ;-) > > > > i've attached my config-files, both ser and openser: > > 21X.XXX.XXX.100 is the host running ser(port 5060) and openser(port 5061) > > 21X.XXX.XXX.101 is asterisk1 > > 21X.XXX.XXX.102 is asterisk2(asterisk2.de.domain.cc) > > > > i'm registering my phones at openser with the domain siptest1.domain.cc, the > > second domain sip2.de.domain.cc is a test for a multidomain setup but doesn't work yet(i can't > > use the domain-column provided by the db since i need two different origination domains in asterisk's sip.conf to seperate the calls - maybe > > not the best approach...) > > > > p.s.: > > a second question regarding the multidomain - i can't register with sip2.de.domain.cc - i guess there's something wrong with my regexp > > > > "if (uri=~"^sip:(.+@)?((sip2\.)?(de\.)?domain\.cc)([:;\?].*)?$") {" > > > > someone having mercy and tell me what is wrong with my syntax(or maybe a > > link to a descriptive documentation - ser admin guide was not enough for > > me)? > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users

2 1

RE: [Users] direct manipulation of DB
by Douglas Garstang 09 Jan '06

09 Jan '06

Isn't there a script called 'sc' somewhere that does the same thing as openserctl? I looked at the sc script and was able to determine how it generates those fields. I think they're gen_ha and something else... installed by openser. -----Original Message----- From: unplug [mailto:maillisting@gmail.com] Sent: Sun 1/8/2006 7:50 PM To: Norman Brandinger Cc: users(a)openser.org Subject: Re: [Users] direct manipulation of DB There are 2 parts. One is server and one is client. How to setup the server as openser has already a fifo interfece? On 12/23/05, Norman Brandinger <norm(a)goes.com> wrote: > I posted a simple FIFO Client that can be used to test sending commands > and receiving responses from OpenSER. > > The link is here: http://openser.org/dokuwiki/doku.php?id=utilities > > Regards, > Norm > > p.s. Comments or suggestions for improvements are welcome. > > p.s.s. I believe that the OpenSER roadmap has an API project that will > depreciate the FIFO interface. Perhaps one of the developers can > elaborate more on this. > > Klaus Darilion wrote: > > > unplug wrote: > > > >> I am using openser with mysql as db. There is a tool call openserctl > >> to manage the data in the openser. I wonder whether I can direct > >> manipulate the db directly without using openserctl. I have tried to > >> insert a record in the aliases table directly using mysql client but > >> it doesn't work. Can I conclude direct manipulate the DB is > >> impossible? > > > > > > It depends on the table! There are some tables which will be cached > > inside openser, e.g: > > location > > aliases > > domain > > gw (lcr module) > > > > If you make changes to location/aliases you have to: > > - restart openser to reload the tables or > > - use openserctl (inserts via FIFO to openser which wirtes back to > > the db or > > - in case of aliases use the aliases_DB module > > > > for the domain and gw table, you can use the respectife FIfo commands > > to reload the tables. > > > > regards > > klaus > > > > _______________________________________________ > > 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

1 0

RE: [Users] Re: OpenSER and transport selection (SRV and NAPTR)
by Douglas Garstang 09 Jan '06

09 Jan '06

Oh yes! That would be AMAZING if that functionality could be added! I think I'm gonna poo my pants! -----Original Message----- From: Greg Fausak [mailto:lgfausak@gmail.com] Sent: Fri 1/6/2006 11:06 AM To: users(a)openser.org Cc: Subject: [Users] Re: OpenSER and transport selection (SRV and NAPTR) Howdy guys, I completely missed this thread. How is this going? I just read the roadmap which indicates NAPTR lookup is on the list. I've been maintaining NAPTR and SRV records looking forward to the day when t_relay() forwards using NAPTR (and SRV). Does that mean that the t_relay() will maintain the state, slogging through the NAPTR/SRV records appropriately, without me having to do additional logic??? Specifically, I'm keen on the SRV serial forking. -g Hi Bogdan, > I meant t_reply() will keep its behaviour as looking into URI for the > destination - but it will incorporate the NAPTR lookup. Great - this answers my question. > to response also to on earlier question, regarding the timing > - I say 3 months are more than enough ;). Excellent. Many thanks for your help (and patience ;) --Joachim > Joachim Fabini wrote: > > >Hi Bogdan, > > > > > > > >>indeed, there was a misunderstanding :) .t_relay() with no > >>param will be kept with the current functionality. > >>Or you suggest to be able to specify only a different proto > >>or port from the RURI? > >> > >> > > > >I did not yet find the primitive which is supposed to > >statefully relay and do the following: > >1. NAPTR query on the transport to use PLUS > >2. _exactly_ what t_relay() does now for the > > transport returned by the NAPTR query. > > > >You say that t_relay() is kept with the current functionality. > >Does this mean no NAPTR, or will t_relay() be extended to use > >NAPTR before SRV/A query? If the latter then everything is OK. > > > >Otherwise we have the alternatives: > > > >t_relay() -> Stateful relaying according to destination > > address using the incoming transport (no NAPTR). > >t_relay_to() -> Stateful relaying to a specific host (you said > > that <host> is mandatory here) using NAPTR to > > determine the transport. > > > > > >To summarize: > >My point is that none of these two primitives covers the case > >when the message is to be relayed to the next hop based only > >on the destination address _and_ using the transport selected > >by the destination proxy (determined via NAPTR query). > > > >Except if either a) t_relay() does NAPTR or b) the host > >parameter in t_relay_to() is optional. > > > >--Joachim > > > > > > > > > -- Greg Fausak greg(a)thursday.com _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

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Re: [Users] deactivate an account
by Helge Waastad 09 Jan '06

09 Jan '06

Hi, In the openserctl script there are acl options static configured (free-pstn etc). Just add inactive on the same line, and it should work br hw There is an error when I issue acl grant openserctl acl grant myname(a)mydomain.com inactive Invalid privilege: inactive ignored -- Mvh/Br Helge Waastad Senior Engineer Systemavdelingen Smartnet

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[Users] direct manipulation of DB
by unplug 09 Jan '06

09 Jan '06

I am using openser with mysql as db. There is a tool call openserctl to manage the data in the openser. I wonder whether I can direct manipulate the db directly without using openserctl. I have tried to insert a record in the aliases table directly using mysql client but it doesn't work. Can I conclude direct manipulate the DB is impossible?

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[Users] Re: Users Digest, Vol 8, Issue 14
by Kenny Yeh 09 Jan '06

09 Jan '06

Hi,Philipp Thanks for your help !!!!!! You show me what I need .thanks!!! ======= 2006-01-07 03:15:05 ��д��======= >Send Users mailing list submissions to > users(a)openser.org > >To subscribe or unsubscribe via the World Wide Web, visit > http://openser.org/cgi-bin/mailman/listinfo/users >or, via email, send a message with subject or body 'help' to > users-request(a)openser.org > >You can reach the person managing the list at > users-owner(a)openser.org > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of Users digest..." > > >Today's Topics: > > 1. uac_replace_from function (unplug) > 2. Re: NAT+uac_replace_from (Daniel-Constantin Mierla) > 3. Re: uac_replace_from function (Daniel-Constantin Mierla) > 4. Re: OpenSER and Solaris (Daniel-Constantin Mierla) > 5. Re: OpenSER and transport selection (SRV and NAPTR) (Greg Fausak) > 6. Re: help: what's tls_ca tls_privat-key tls_calist > (Alexander Philipp Lintenhofer) > 7. contact field port problem when behind nats (Jack Wei) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Fri, 6 Jan 2006 19:08:34 +0800 >From: unplug <maillisting(a)gmail.com> >Subject: [Users] uac_replace_from function >To: users(a)openser.org >Message-ID: > <6fbb529e0601060308p5a0eab32wafce0fcc5f04b286(a)mail.gmail.com> >Content-Type: text/plain; charset=ISO-8859-1 > >I am using uac_replace_from function as below. > ># replace only display and do not touch uri >uac_replace_from("35418474",""); > >After running uac_replace_from function, I get the following message. > >U 203.193.4.242:5060 -> 203.193.4.226:5060 >SIP/2.0 183 Session Progress. >Via: SIP/2.0/UDP >203.193.4.226;branch=z9hG4bKaed7.46cc4101.0,SIP/2.0/UDP >10.0.0.52:5060;rport=5060;received=210.184.4.31;branch=z9hG4bKHNHpldVsKca6IYUS. >From: "35418474"<sip:882754853589@ow01.owtel.com>;tag=sBl2gLH6O1myyqL5. >To: "92588819" <sip:92588819@ow01.owtel.com>;tag=47C61DF4-26CD. >Date: Fri, 06 Jan 2006 10:41:17 GMT. >Call-ID: yTCQvS56ZCZXCHgt(a)10.0.0.52. >Server: Cisco-SIPGateway/IOS-12.x. > >In the normal message, sip message, there is a space between display >name and uri. >ie. From: "35418474" <sip:882754853589@ow01.owtel.com> >However, after I apply uac_replace_from function, there is a space >there. How can I add a space such that it looks like a normal sip >message? Does the space will affect the caller display? > > > >------------------------------ > >Message: 2 >Date: Fri, 06 Jan 2006 13:28:52 +0200 >From: Daniel-Constantin Mierla <daniel(a)voice-system.ro> >Subject: Re: [Users] NAT+uac_replace_from >To: unplug <maillisting(a)gmail.com> >Cc: users(a)openser.org >Message-ID: <43BE5474.4010400(a)voice-system.ro> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >Send me the network traces for this case to see what is wrong. The uac >module has a mode when it should do automatic replacement in subsequent >requests. > >Daniel > > >On 01/06/06 04:37, unplug wrote: >> Thanks for your reply. According to your reply, I add the following >> code for the PRACK. >> if (method=="PRACK") { >> if (avp_db_load("$from/uri","s:alias")) { >> xlog("L_INFO","sip408: have alias - [$avp(s:alias)]\n"); >> uac_replace_from("anonymous","sip:$avp(s:alias)@$si"); >> } else { >> xlog("L_INFO","sip411: no alias\n"); >> }; >> } >> >> The result is the same as before. It seems there is a conflict in the >> from header and it causes the 481 error. To be more precise, below is >> the protocol flow. >> >> UA1 server1 server2 >> fromtag:taga -----from username:9000 INVITE-----> >> fromtag:taga <----from username:9000 100 trying--- >> after uac_replace_from("1234","sip:1234@$si") >> fromtag:taga ------from username:1234 INVITE ----> >> (callee rings but caller drops) >> fromtag:taga <-----from username:1234 100 trying---- >> (callee still rings until answers the call) >> fromtag:taga <----from username:1234 session progress 183--- >> fromtag:taga <-----from username:9000 session prgress 183---- >> fromtag:taga -----from username:9000 PRACK-----------> >> fromtag:taga ----------from username:9000 PRACK---> >> fromtag:taga <----from username:9000 481 call leg ----- >> fromtag:taga <----from username:9000 481 call leg ----- >> >> >> On 1/5/06, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: >> >>> Hello, >>> >>> "SIP/2.0 481 Call Leg/Transaction Does Not Exist" is for PRACK, because you do not change the From header for it. You have to do the same translation for all requests within the dialog. See the documentation of uac module: . >>> >>> http://openser.org/docs/modules/1.1.x/uac.html >>> >>> Cheers, >>> Daniel >>> >>> >>> On 01/04/06 12:21, unplug wrote: >>> >>>> Actually, I am replacing the username of the uri with the alias that >>>> stored in the database. In my configuration file, it is using >>>> mediaproxy for NAT function (features-callfwd.5.0.cfg from getting >>>> started). I also add the following codes in the very first of the >>>> route routine for alias replacing purpose. >>>> >>>> route { >>>> ... >>>> if (!has_totag() && method=="INVITE") { >>>> if (avp_db_load("$from/uri","s:alias")) { >>>> xlog("L_INFO","sip408: have alias - [$avp(s:alias)]\n"); >>>> uac_replace_from("anonymous","sip:$avp(s:alias)@$si"); >>>> } else { >>>> xlog("L_INFO","sip411: no alias\n"); >>>> }; >>>> }; >>>> ... >>>> } >>>> >>>> When I make a call from a phone to the PSTN phone, the caller drops >>>> the ring when the callee rings. Callee hangs up and callee rings >>>> again few seconds after. You can find an error message "SIP/2.0 481 >>>> Call Leg/Transaction Does Not Exist" in line 180. I wonder if there >>>> is any thing wrong with the above code or my concept is wrong. Please >>>> help. Below is the sip message. >>>> [...] >>>> >>>> >> >> > > > >------------------------------ > >Message: 3 >Date: Fri, 06 Jan 2006 13:33:32 +0200 >From: Daniel-Constantin Mierla <daniel(a)voice-system.ro> >Subject: Re: [Users] uac_replace_from function >To: unplug <maillisting(a)gmail.com> >Cc: users(a)openser.org >Message-ID: <43BE558C.70404(a)voice-system.ro> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >It might be a bug in the module, please send the original SIP reply as >well, to be able to trace the problem quicker. > >Daniel > > >On 01/06/06 13:08, unplug wrote: >> I am using uac_replace_from function as below. >> >> # replace only display and do not touch uri >> uac_replace_from("35418474",""); >> >> After running uac_replace_from function, I get the following message. >> >> U 203.193.4.242:5060 -> 203.193.4.226:5060 >> SIP/2.0 183 Session Progress. >> Via: SIP/2.0/UDP >> 203.193.4.226;branch=z9hG4bKaed7.46cc4101.0,SIP/2.0/UDP >> 10.0.0.52:5060;rport=5060;received=210.184.4.31;branch=z9hG4bKHNHpldVsKca6IYUS. >> From: "35418474"<sip:882754853589@ow01.owtel.com>;tag=sBl2gLH6O1myyqL5. >> To: "92588819" <sip:92588819@ow01.owtel.com>;tag=47C61DF4-26CD. >> Date: Fri, 06 Jan 2006 10:41:17 GMT. >> Call-ID: yTCQvS56ZCZXCHgt(a)10.0.0.52. >> Server: Cisco-SIPGateway/IOS-12.x. >> >> In the normal message, sip message, there is a space between display >> name and uri. >> ie. From: "35418474" <sip:882754853589@ow01.owtel.com> >> However, after I apply uac_replace_from function, there is a space >> there. How can I add a space such that it looks like a normal sip >> message? Does the space will affect the caller display? >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > > >------------------------------ > >Message: 4 >Date: Fri, 06 Jan 2006 19:41:19 +0200 >From: Daniel-Constantin Mierla <daniel(a)voice-system.ro> >Subject: Re: [Users] OpenSER and Solaris >To: martin(a)campus-merseburg.de >Cc: users(a)openser.org >Message-ID: <43BEABBF.3090302(a)voice-system.ro> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >Hello, > >thanks for reporting, I have renamed the internal structure meminfo to >mem_info to avoid this conflict. Please take the latest 1.0.0 from cvs >and let me know if it works. > >Here are the commands to get the stable version from cvs: > ># export CVSROOT=:pserver:anonymous@cvs.sourceforge.net:/cvsroot/openser ># cvs login - for password simply press the Enter key. ># cvs -z3 checkout -r rel_1_0_0 sip-server > > >Cheers, >Daniel > > >On 01/06/06 12:45, martin(a)campus-merseburg.de wrote: >> Hi, >> >> does anyone use OpenSER under Solaris? >> >> I get this error when compiling: >> >> gcc -g -O9 -funroll-loops -Wall -mcpu=ultrasparc -mtune=ultrasparc >> -DNAME='"openser"' -DVERSION='"1.0.0"' -DARCH='"sparc64"' >> -DOS='"solaris"' -DCOMPILER='"gcc 3.4"' -D__CPU_sparc64 >> -D__OS_solaris -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC >> -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP >> -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DFAST_LOCK >> -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETIPNODEBYNAME >> -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -c action.c -o >> action.o >> In file included from mem/f_malloc.h:40, >> from mem/mem.h:54, >> from action.c:50: >> mem/meminfo.h:33: error: redefinition of `struct meminfo' >> gmake: *** [action.o] Error 1 >> >> any idea for fixing this? >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > > >------------------------------ > >Message: 5 >Date: Fri, 6 Jan 2006 12:06:42 -0600 >From: Greg Fausak <lgfausak(a)gmail.com> >Subject: [Users] Re: OpenSER and transport selection (SRV and NAPTR) >To: users(a)openser.org >Message-ID: > <8c7b802f0601061006p373d8949we5e48c9953eb8dc9(a)mail.gmail.com> >Content-Type: text/plain; charset=ISO-8859-1 > >Howdy guys, > >I completely missed this thread. >How is this going? > >I just read the roadmap which indicates NAPTR lookup >is on the list. I've been maintaining NAPTR and SRV records >looking forward to the day when t_relay() forwards using NAPTR (and >SRV). Does that mean that the t_relay() will maintain the state, slogging >through the NAPTR/SRV records appropriately, without me having >to do additional logic??? Specifically, I'm keen on the SRV serial forking. > >-g > > > > >Hi Bogdan, > >> I meant t_reply() will keep its behaviour as looking into URI for the >> destination - but it will incorporate the NAPTR lookup. > >Great - this answers my question. > > >> to response also to on earlier question, regarding the timing >> - I say 3 months are more than enough ;). > >Excellent. > >Many thanks for your help (and patience ;) >--Joachim > > > >> Joachim Fabini wrote: >> >> >Hi Bogdan, >> > >> > >> > >> >>indeed, there was a misunderstanding :) .t_relay() with no >> >>param will be kept with the current functionality. >> >>Or you suggest to be able to specify only a different proto >> >>or port from the RURI? >> >> >> >> >> > >> >I did not yet find the primitive which is supposed to >> >statefully relay and do the following: >> >1. NAPTR query on the transport to use PLUS >> >2. _exactly_ what t_relay() does now for the >> > transport returned by the NAPTR query. >> > >> >You say that t_relay() is kept with the current functionality. >> >Does this mean no NAPTR, or will t_relay() be extended to use >> >NAPTR before SRV/A query? If the latter then everything is OK. >> > >> >Otherwise we have the alternatives: >> > >> >t_relay() -> Stateful relaying according to destination >> > address using the incoming transport (no NAPTR). >> >t_relay_to() -> Stateful relaying to a specific host (you said >> > that <host> is mandatory here) using NAPTR to >> > determine the transport. >> > >> > >> >To summarize: >> >My point is that none of these two primitives covers the case >> >when the message is to be relayed to the next hop based only >> >on the destination address _and_ using the transport selected >> >by the destination proxy (determined via NAPTR query). >> > >> >Except if either a) t_relay() does NAPTR or b) the host >> >parameter in t_relay_to() is optional. >> > >> >--Joachim >> > >> > >> > >> > >> >-- >Greg Fausak >greg(a)thursday.com > > > >------------------------------ > >Message: 6 >Date: Fri, 06 Jan 2006 19:47:10 +0100 >From: Alexander Philipp Lintenhofer <lintenhofer(a)aon.at> >Subject: Re: [Users] help: what's tls_ca tls_privat-key tls_calist >To: users(a)openser.org >Message-ID: <20060106194710.x2agnokfltq8ok4s(a)liho.dyndns.org> >Content-Type: text/plain; charset=UTF-8; format="flowed" > >Hi Kenny, > >It would be a big deal first to acquire knowledge about the theoretical >background before installing this service. > >So please read following (and related) sites: >http://www.tech-invite.com/Ti-sip-security.html >http://www.tech-invite.com/Ti-SSL.html > >The definitive documentation about OpenSER/TLS >(http://openser.org/docs/tls.html) is very good and therefore easy to >understand in succession! > >regards, >Philipp > >> Hello all, >> >> I have tested tls with openser by SIPp,but I donn't understant >> the flows of TLS ctreated,who can show me the flow about TLS >> handshake. >> >> When I test tls by SIPp,I put the ca,private-key to it's >> DIR,that's fils created by openssl ,and add these files path to cfg >> of openser.the same files used.I want to know what's mean is tls_ca >> tls_privatekey tls_calist,and how there files work. Client need >> what's ca and private-key for creating tls connection? Is it the same >> with openssl? to create ssl connection? >> >> >> >> >> >> ======= 2006-01-05 19:01:27 您在来信中写道：======= >> >>> Send Users mailing list submissions to >>> users(a)openser.org >>> >>> To subscribe or unsubscribe via the World Wide Web, visit >>> http://openser.org/cgi-bin/mailman/listinfo/users >>> or, via email, send a message with subject or body 'help' to >>> users-request(a)openser.org >>> >>> You can reach the person managing the list at >>> users-owner(a)openser.org >>> >>> When replying, please edit your Subject line so it is more specific >>> than "Re: Contents of Users digest..." >>> >>> >>> Today's Topics: >>> >>> 1. Re: deactivate an account (Helge Waastad) >>> >>> >>> ---------------------------------------------------------------------- >>> >>> Message: 1 >>> Date: Thu, 05 Jan 2006 11:23:30 +0100 >>> From: Helge Waastad <helge(a)smartnet.no> >>> Subject: Re: [Users] deactivate an account >>> To: maillisting(a)gmail.com >>> Cc: users(a)openser.org >>> Message-ID: <1136456610.11157.10.camel(a)localhost.localdomain> >>> Content-Type: text/plain >>> >>> Hi, >>> In the openserctl script there are acl options static configured >>> (free-pstn etc). >>> Just add inactive on the same line, and it should work >>> >>> br hw >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> There is an error when I issue acl grant >>> openserctl acl grant myname(a)mydomain.com inactive >>> Invalid privilege: inactive ignored >>> >>> >>> >>> -- >>> Mvh/Br >>> Helge Waastad >>> Senior Engineer >>> Systemavdelingen >>> Smartnet >>> >>> >>> >>> ------------------------------ >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)openser.org >>> http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >>> End of Users Digest, Vol 8, Issue 11 >>> ************************************ >> >> = = = = = = = = = = = = = = = = = = = = >> >> >> 　　　　　　　　�? >> 礼！ >> >> >> 　　　　　　　　Kenny Yeh >> 　　　　　　　　kenny(a)artdio.com.tw >> 　　　　　　　　　　2006-01-06 >> >> > > > > > > >------------------------------ > >Message: 7 >Date: Fri, 6 Jan 2006 11:14:14 -0800 (PST) >From: Jack Wei <cowlemon(a)yahoo.com> >Subject: [Users] contact field port problem when behind nats >To: users(a)openser.org >Message-ID: <20060106191415.56473.qmail(a)web50008.mail.yahoo.com> >Content-Type: text/plain; charset=iso-8859-1 > >Hi, > >I'm having a problem when an UAC is behind NAT and has a public port other than >5060. The public port isn't present in the contact field so when an ACK for an >INVITE is sent to the UAC, it's directed to the default port of 5060. I'm >currently using v1.0.0 w/ the NATHELPER module. The UAC is SJphone. Any help >would be greatly appreciated. > > > > > >NAT part of my openser.cfg : > >if (nat_uac_test("19")) >{ > if ( method == "REGISTER" ) > { > fix_nated_register(); > force_rport(); > } > else > { > force_rport(); > fix_nated_contact(); > > if (method == "INVITE") > { > fix_nated_sdp("1"); > }; > }; > > append_hf( "P-hint: NAT traversal\r\n" ); >} > > > > >the sip proxy is voip.somewhere.com and 1.1.1.1 >the pstn gateway is 2.2.2.2 > >the caller is from pstn and the callee is the uac. > > >Here's a log of SIP msgs: > ># >U 2.2.2.2:5060 -> 1.1.1.1:5060 >INVITE sip:2139435286@voip.somewhere.com:5060;user=phone SIP/2.0. >t: <sip:2139435286@voip.somewhere.com:5060;user=phone>. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >Remote-Party-Id: ><sip:2139961998@2.2.2.2:5060;user=phone>;screen=no;id-type=subscriber;party=calling;privacy=off. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 INVITE. >v: SIP/2.0/UDP 2.2.2.2:5060. >Max-Forwards: 70. >m: <sip:2139961998@2.2.2.2:5060;user=phone>. >k: replaces. >c: application/sdp. >Accept: application/sdp. >Accept-Encoding: . >Accept-Language: en. >User-Agent: Lucent-Universal-Gateway. >l: 244. >. >v=0. >o=VTX-SIP-TNT01 505428282 505428282 IN IP4 2.2.2.2. >s=Session SDP. >c=IN IP4 2.2.2.2. >t=0 0. >m=audio 41788 RTP/AVP 0 101. >a=silenceSupp:off. >a=ecan:b on g168. >a=ptime:20. >a=rtpmap:101 telephone-event/8000. >a=rtpmap:0 PCMU/8000. > ># >U 1.1.1.1:5060 -> 2.2.2.2:5060 >SIP/2.0 100 trying -- your call is important to us. >t: <sip:2139435286@voip.somewhere.com:5060;user=phone>. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 INVITE. >v: SIP/2.0/UDP 2.2.2.2:5060. >Server: OpenSer (1.0.0-tls (i386/linux)). >Content-Length: 0. >. > ># >U 1.1.1.1:5060 -> 64.77.236.227:32883 >INVITE sip:2139435286@64.77.236.227 SIP/2.0. >Record-Route: ><sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. >t: <sip:2139435286@voip.somewhere.com:5060;user=phone>. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >Remote-Party-Id: ><sip:2139961998@2.2.2.2:5060;user=phone>;screen=no;id-type=subscriber;party=calling;privacy=off. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 INVITE. >Via: SIP/2.0/UDP voip.somewhere.com;branch=z9hG4bK8325.35c96c63.0. >v: SIP/2.0/UDP 2.2.2.2:5060. >Max-Forwards: 69. >m: <sip:2139961998@2.2.2.2:5060;user=phone>. >k: replaces. >c: application/sdp. >Accept: application/sdp. >Accept-Encoding: . >Accept-Language: en. >User-Agent: Lucent-Universal-Gateway. >l: 244. >P-hint: VOIP OUTBOUND. >. >v=0. >o=VTX-SIP-TNT01 505428282 505428282 IN IP4 2.2.2.2. >s=Session SDP. >c=IN IP4 2.2.2.2. >t=0 0. >m=audio 41788 RTP/AVP 0 101. >a=silenceSupp:off. >a=ecan:b on g168. >a=ptime:20. >a=rtpmap:101 telephone-event/8000. >a=rtpmap:0 PCMU/8000. > ># >U 64.77.236.227:32883 -> 1.1.1.1:5060 >SIP/2.0 100 Trying. >v: SIP/2.0/UDP >voip.somewhere.com;received=1.1.1.1;branch=z9hG4bK8325.35c96c63.0,SIP/2.0/UDP >2.2.2.2:5060. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >t: ""<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 INVITE. >l: 0. >Server: SJphone/1.61.306c (SJ Labs). >. > ># >U 64.77.236.227:32883 -> 1.1.1.1:5060 >SIP/2.0 200 OK. >v: SIP/2.0/UDP >voip.somewhere.com;received=1.1.1.1;branch=z9hG4bK8325.35c96c63.0,SIP/2.0/UDP >2.2.2.2:5060. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >t: "Jack >Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >m: <sip:2139435286@64.77.236.227>. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 INVITE. >l: 217. >c: application/sdp. >Server: SJphone/1.61.306c (SJ Labs). >Record-Route: ><sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. >. >v=0. >o=- 3345559486 3345559486 IN IP4 64.77.236.227. >s=SJphone. >c=IN IP4 64.77.236.227. >t=0 0. >a=setup:active. >m=audio 49154 RTP/AVP 0 101. >a=rtpmap:0 PCMU/8000. >a=rtpmap:101 telephone-event/8000. >a=fmtp:101 0-11,16. > ># >U 1.1.1.1:5060 -> 2.2.2.2:5060 >SIP/2.0 200 OK. >v: SIP/2.0/UDP 2.2.2.2:5060. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >t: "Jack >Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >m: <sip:2139435286@64.77.236.227>. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 INVITE. >l: 217. >c: application/sdp. >Server: SJphone/1.61.306c (SJ Labs). >Record-Route: ><sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. >. >v=0. >o=- 3345559486 3345559486 IN IP4 64.77.236.227. >s=SJphone. >c=IN IP4 64.77.236.227. >t=0 0. >a=setup:active. >m=audio 49154 RTP/AVP 0 101. >a=rtpmap:0 PCMU/8000. >a=rtpmap:101 telephone-event/8000. >a=fmtp:101 0-11,16. > ># >U 2.2.2.2:5060 -> 1.1.1.1:5060 >ACK sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on >SIP/2.0. >t: <sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 ACK. >v: SIP/2.0/UDP 2.2.2.2:5060. >Max-Forwards: 70. >Route: <sip:2139435286@64.77.236.227>. >User-Agent: Lucent-Universal-Gateway. >l: 0. >. > ># >U 1.1.1.1:5060 -> 64.77.236.227:5060 >ACK sip:2139435286@64.77.236.227 SIP/2.0. >Record-Route: ><sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. >t: <sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 ACK. >Via: SIP/2.0/UDP voip.somewhere.com;branch=0. >v: SIP/2.0/UDP 2.2.2.2:5060. >Max-Forwards: 69. >User-Agent: Lucent-Universal-Gateway. >l: 0. >P-hint: rr-enforced. >. > ># >U 64.77.236.227:32883 -> 1.1.1.1:5060 >SIP/2.0 200 OK. >v: SIP/2.0/UDP >voip.somewhere.com;received=1.1.1.1;branch=z9hG4bK8325.35c96c63.0,SIP/2.0/UDP >2.2.2.2:5060. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >t: "Jack >Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >m: <sip:2139435286@64.77.236.227>. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 INVITE. >l: 217. >c: application/sdp. >Server: SJphone/1.61.306c (SJ Labs). >Record-Route: ><sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. >. >v=0. >o=- 3345559486 3345559486 IN IP4 64.77.236.227. >s=SJphone. >c=IN IP4 64.77.236.227. >t=0 0. >a=setup:active. >m=audio 49154 RTP/AVP 0 101. >a=rtpmap:0 PCMU/8000. >a=rtpmap:101 telephone-event/8000. >a=fmtp:101 0-11,16. > ># >U 1.1.1.1:5060 -> 2.2.2.2:5060 >SIP/2.0 200 OK. >v: SIP/2.0/UDP 2.2.2.2:5060. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >t: "Jack >Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >m: <sip:2139435286@64.77.236.227>. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 INVITE. >l: 217. >c: application/sdp. >Server: SJphone/1.61.306c (SJ Labs). >Record-Route: ><sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. >. >v=0. >o=- 3345559486 3345559486 IN IP4 64.77.236.227. >s=SJphone. >c=IN IP4 64.77.236.227. >t=0 0. >a=setup:active. >m=audio 49154 RTP/AVP 0 101. >a=rtpmap:0 PCMU/8000. >a=rtpmap:101 telephone-event/8000. >a=fmtp:101 0-11,16. > ># >U 2.2.2.2:5060 -> 1.1.1.1:5060 >ACK sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on >SIP/2.0. >t: <sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 ACK. >v: SIP/2.0/UDP 2.2.2.2:5060. >Max-Forwards: 70. >Route: <sip:2139435286@64.77.236.227>. >User-Agent: Lucent-Universal-Gateway. >l: 0. >. > ># >U 1.1.1.1:5060 -> 64.77.236.227:5060 >ACK sip:2139435286@64.77.236.227 SIP/2.0. >Record-Route: ><sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. >t: <sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 7528304 ACK. >Via: SIP/2.0/UDP voip.somewhere.com;branch=0. >v: SIP/2.0/UDP 2.2.2.2:5060. >Max-Forwards: 69. >User-Agent: Lucent-Universal-Gateway. >l: 0. >P-hint: rr-enforced. >. > > > > > > >................<REPEAT OKs and ACKs>........................ > > > > > > > > ># >U 64.77.236.227:32883 -> 1.1.1.1:5060 >BYE sip:2139961998@2.2.2.2:5060;user=phone SIP/2.0. >v: SIP/2.0/UDP >192.168.0.100;rport;branch=z9hG4bKc0a800640000002243beb15e0000571700000028. >f: "Jack >Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >t: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 1 BYE. >Max-Forwards: 70. >User-Agent: SJphone/1.61.306c (SJ Labs). >l: 0. >Route: ><sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. >. > ># >U 1.1.1.1:5060 -> 2.2.2.2:5060 >BYE sip:2139961998@2.2.2.2:5060;user=phone SIP/2.0. >Record-Route: <sip:2139961998@voip.somewhere.com;ftag=71451719939;lr=on>. >Via: SIP/2.0/UDP voip.somewhere.com;branch=z9hG4bK4f59.28705173.0. >v: SIP/2.0/UDP >192.168.0.100;received=64.77.236.227;rport=32883;branch=z9hG4bKc0a800640000002243beb15e0000571700000028. >f: "Jack >Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >t: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 1 BYE. >Max-Forwards: 69. >User-Agent: SJphone/1.61.306c (SJ Labs). >l: 0. >P-hint: NAT traversal. >P-hint: rr-enforced. >. > ># >U 2.2.2.2:5060 -> 1.1.1.1:5060 >SIP/2.0 200 OK. >v: SIP/2.0/UDP voip.somewhere.com;branch=z9hG4bK4f59.28705173.0. >v: SIP/2.0/UDP >192.168.0.100;received=64.77.236.227;rport=32883;branch=z9hG4bKc0a800640000002243beb15e0000571700000028. >t: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >f: "Jack Wei" ><sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 1 BYE. >Server: Lucent-Universal-Gateway. >User-Agent: Lucent-Universal-Gateway. >l: 0. >. > ># >U 1.1.1.1:5060 -> 64.77.236.227:32883 >SIP/2.0 200 OK. >v: SIP/2.0/UDP >192.168.0.100;received=64.77.236.227;rport=32883;branch=z9hG4bKc0a800640000002243beb15e0000571700000028. >t: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. >f: "Jack Wei" ><sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. >i: dfc2b0e2-380-1e20393a(a)2.2.2.2. >CSeq: 1 BYE. >Server: Lucent-Universal-Gateway. >User-Agent: Lucent-Universal-Gateway. >l: 0. >. > > >Jack Wei > > > >__________________________________________ >Yahoo! DSL ?Something to write home about. >Just $16.99/mo. or less. >dsl.yahoo.com > > > > >------------------------------ > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > >End of Users Digest, Vol 8, Issue 14 >************************************ = = = = = = = = = = = = = = = = = = = = �� Kenny Yeh ��kenny(a)artdio.com.tw ��2006-01-09

1 0

[Users] contact field port problem when behind nats
by Jack Wei 07 Jan '06

07 Jan '06

Hi, I'm having a problem when an UAC is behind NAT and has a public port other than 5060. The public port isn't present in the contact field so when an ACK for an INVITE is sent to the UAC, it's directed to the default port of 5060. I'm currently using v1.0.0 w/ the NATHELPER module. The UAC is SJphone. Any help would be greatly appreciated. NAT part of my openser.cfg : if (nat_uac_test("19")) { if ( method == "REGISTER" ) { fix_nated_register(); force_rport(); } else { force_rport(); fix_nated_contact(); if (method == "INVITE") { fix_nated_sdp("1"); }; }; append_hf( "P-hint: NAT traversal\r\n" ); } the sip proxy is voip.somewhere.com and 1.1.1.1 the pstn gateway is 2.2.2.2 the caller is from pstn and the callee is the uac. Here's a log of SIP msgs: # U 2.2.2.2:5060 -> 1.1.1.1:5060 INVITE sip:2139435286@voip.somewhere.com:5060;user=phone SIP/2.0. t: <sip:2139435286@voip.somewhere.com:5060;user=phone>. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. Remote-Party-Id: <sip:2139961998@2.2.2.2:5060;user=phone>;screen=no;id-type=subscriber;party=calling;privacy=off. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 INVITE. v: SIP/2.0/UDP 2.2.2.2:5060. Max-Forwards: 70. m: <sip:2139961998@2.2.2.2:5060;user=phone>. k: replaces. c: application/sdp. Accept: application/sdp. Accept-Encoding: . Accept-Language: en. User-Agent: Lucent-Universal-Gateway. l: 244. . v=0. o=VTX-SIP-TNT01 505428282 505428282 IN IP4 2.2.2.2. s=Session SDP. c=IN IP4 2.2.2.2. t=0 0. m=audio 41788 RTP/AVP 0 101. a=silenceSupp:off. a=ecan:b on g168. a=ptime:20. a=rtpmap:101 telephone-event/8000. a=rtpmap:0 PCMU/8000. # U 1.1.1.1:5060 -> 2.2.2.2:5060 SIP/2.0 100 trying -- your call is important to us. t: <sip:2139435286@voip.somewhere.com:5060;user=phone>. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 INVITE. v: SIP/2.0/UDP 2.2.2.2:5060. Server: OpenSer (1.0.0-tls (i386/linux)). Content-Length: 0. . # U 1.1.1.1:5060 -> 64.77.236.227:32883 INVITE sip:2139435286@64.77.236.227 SIP/2.0. Record-Route: <sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. t: <sip:2139435286@voip.somewhere.com:5060;user=phone>. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. Remote-Party-Id: <sip:2139961998@2.2.2.2:5060;user=phone>;screen=no;id-type=subscriber;party=calling;privacy=off. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 INVITE. Via: SIP/2.0/UDP voip.somewhere.com;branch=z9hG4bK8325.35c96c63.0. v: SIP/2.0/UDP 2.2.2.2:5060. Max-Forwards: 69. m: <sip:2139961998@2.2.2.2:5060;user=phone>. k: replaces. c: application/sdp. Accept: application/sdp. Accept-Encoding: . Accept-Language: en. User-Agent: Lucent-Universal-Gateway. l: 244. P-hint: VOIP OUTBOUND. . v=0. o=VTX-SIP-TNT01 505428282 505428282 IN IP4 2.2.2.2. s=Session SDP. c=IN IP4 2.2.2.2. t=0 0. m=audio 41788 RTP/AVP 0 101. a=silenceSupp:off. a=ecan:b on g168. a=ptime:20. a=rtpmap:101 telephone-event/8000. a=rtpmap:0 PCMU/8000. # U 64.77.236.227:32883 -> 1.1.1.1:5060 SIP/2.0 100 Trying. v: SIP/2.0/UDP voip.somewhere.com;received=1.1.1.1;branch=z9hG4bK8325.35c96c63.0,SIP/2.0/UDP 2.2.2.2:5060. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. t: ""<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 INVITE. l: 0. Server: SJphone/1.61.306c (SJ Labs). . # U 64.77.236.227:32883 -> 1.1.1.1:5060 SIP/2.0 200 OK. v: SIP/2.0/UDP voip.somewhere.com;received=1.1.1.1;branch=z9hG4bK8325.35c96c63.0,SIP/2.0/UDP 2.2.2.2:5060. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. t: "Jack Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. m: <sip:2139435286@64.77.236.227>. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 INVITE. l: 217. c: application/sdp. Server: SJphone/1.61.306c (SJ Labs). Record-Route: <sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. . v=0. o=- 3345559486 3345559486 IN IP4 64.77.236.227. s=SJphone. c=IN IP4 64.77.236.227. t=0 0. a=setup:active. m=audio 49154 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11,16. # U 1.1.1.1:5060 -> 2.2.2.2:5060 SIP/2.0 200 OK. v: SIP/2.0/UDP 2.2.2.2:5060. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. t: "Jack Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. m: <sip:2139435286@64.77.236.227>. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 INVITE. l: 217. c: application/sdp. Server: SJphone/1.61.306c (SJ Labs). Record-Route: <sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. . v=0. o=- 3345559486 3345559486 IN IP4 64.77.236.227. s=SJphone. c=IN IP4 64.77.236.227. t=0 0. a=setup:active. m=audio 49154 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11,16. # U 2.2.2.2:5060 -> 1.1.1.1:5060 ACK sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on SIP/2.0. t: <sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 ACK. v: SIP/2.0/UDP 2.2.2.2:5060. Max-Forwards: 70. Route: <sip:2139435286@64.77.236.227>. User-Agent: Lucent-Universal-Gateway. l: 0. . # U 1.1.1.1:5060 -> 64.77.236.227:5060 ACK sip:2139435286@64.77.236.227 SIP/2.0. Record-Route: <sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. t: <sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 ACK. Via: SIP/2.0/UDP voip.somewhere.com;branch=0. v: SIP/2.0/UDP 2.2.2.2:5060. Max-Forwards: 69. User-Agent: Lucent-Universal-Gateway. l: 0. P-hint: rr-enforced. . # U 64.77.236.227:32883 -> 1.1.1.1:5060 SIP/2.0 200 OK. v: SIP/2.0/UDP voip.somewhere.com;received=1.1.1.1;branch=z9hG4bK8325.35c96c63.0,SIP/2.0/UDP 2.2.2.2:5060. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. t: "Jack Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. m: <sip:2139435286@64.77.236.227>. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 INVITE. l: 217. c: application/sdp. Server: SJphone/1.61.306c (SJ Labs). Record-Route: <sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. . v=0. o=- 3345559486 3345559486 IN IP4 64.77.236.227. s=SJphone. c=IN IP4 64.77.236.227. t=0 0. a=setup:active. m=audio 49154 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11,16. # U 1.1.1.1:5060 -> 2.2.2.2:5060 SIP/2.0 200 OK. v: SIP/2.0/UDP 2.2.2.2:5060. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. t: "Jack Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. m: <sip:2139435286@64.77.236.227>. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 INVITE. l: 217. c: application/sdp. Server: SJphone/1.61.306c (SJ Labs). Record-Route: <sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. . v=0. o=- 3345559486 3345559486 IN IP4 64.77.236.227. s=SJphone. c=IN IP4 64.77.236.227. t=0 0. a=setup:active. m=audio 49154 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11,16. # U 2.2.2.2:5060 -> 1.1.1.1:5060 ACK sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on SIP/2.0. t: <sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 ACK. v: SIP/2.0/UDP 2.2.2.2:5060. Max-Forwards: 70. Route: <sip:2139435286@64.77.236.227>. User-Agent: Lucent-Universal-Gateway. l: 0. . # U 1.1.1.1:5060 -> 64.77.236.227:5060 ACK sip:2139435286@64.77.236.227 SIP/2.0. Record-Route: <sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. t: <sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. f: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 7528304 ACK. Via: SIP/2.0/UDP voip.somewhere.com;branch=0. v: SIP/2.0/UDP 2.2.2.2:5060. Max-Forwards: 69. User-Agent: Lucent-Universal-Gateway. l: 0. P-hint: rr-enforced. . ................<REPEAT OKs and ACKs>........................ # U 64.77.236.227:32883 -> 1.1.1.1:5060 BYE sip:2139961998@2.2.2.2:5060;user=phone SIP/2.0. v: SIP/2.0/UDP 192.168.0.100;rport;branch=z9hG4bKc0a800640000002243beb15e0000571700000028. f: "Jack Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. t: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 1 BYE. Max-Forwards: 70. User-Agent: SJphone/1.61.306c (SJ Labs). l: 0. Route: <sip:2139435286@voip.somewhere.com;ftag=20712b63-1e20393a-83ef4d40;lr=on>. . # U 1.1.1.1:5060 -> 2.2.2.2:5060 BYE sip:2139961998@2.2.2.2:5060;user=phone SIP/2.0. Record-Route: <sip:2139961998@voip.somewhere.com;ftag=71451719939;lr=on>. Via: SIP/2.0/UDP voip.somewhere.com;branch=z9hG4bK4f59.28705173.0. v: SIP/2.0/UDP 192.168.0.100;received=64.77.236.227;rport=32883;branch=z9hG4bKc0a800640000002243beb15e0000571700000028. f: "Jack Wei"<sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. t: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 1 BYE. Max-Forwards: 69. User-Agent: SJphone/1.61.306c (SJ Labs). l: 0. P-hint: NAT traversal. P-hint: rr-enforced. . # U 2.2.2.2:5060 -> 1.1.1.1:5060 SIP/2.0 200 OK. v: SIP/2.0/UDP voip.somewhere.com;branch=z9hG4bK4f59.28705173.0. v: SIP/2.0/UDP 192.168.0.100;received=64.77.236.227;rport=32883;branch=z9hG4bKc0a800640000002243beb15e0000571700000028. t: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. f: "Jack Wei" <sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 1 BYE. Server: Lucent-Universal-Gateway. User-Agent: Lucent-Universal-Gateway. l: 0. . # U 1.1.1.1:5060 -> 64.77.236.227:32883 SIP/2.0 200 OK. v: SIP/2.0/UDP 192.168.0.100;received=64.77.236.227;rport=32883;branch=z9hG4bKc0a800640000002243beb15e0000571700000028. t: <sip:2139961998@2.2.2.2:5060;user=phone>;tag=20712b63-1e20393a-83ef4d40. f: "Jack Wei" <sip:2139435286@voip.somewhere.com:5060;user=phone>;tag=71451719939. i: dfc2b0e2-380-1e20393a(a)2.2.2.2. CSeq: 1 BYE. Server: Lucent-Universal-Gateway. User-Agent: Lucent-Universal-Gateway. l: 0. . Jack Wei __________________________________________ Yahoo! DSL Something to write home about. Just $16.99/mo. or less. dsl.yahoo.com

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[Users] help: what's tls_ca tls_privat-key tls_calist
by Kenny Yeh 06 Jan '06

06 Jan '06

Hello all, I have tested tls with openser by SIPp,but I donn't understant the flows of TLS ctreated,who can show me the flow about TLS handshake. When I test tls by SIPp,I put the ca,private-key to it's DIR,that's fils created by openssl ,and add these files path to cfg of openser.the same files used.I want to know what's mean is tls_ca tls_privatekey tls_calist,and how there files work. Client need what's ca and private-key for creating tls connection? Is it the same with openssl? to create ssl connection? ======= 2006-01-05 19:01:27 您在来信中写道：======= >Send Users mailing list submissions to > users(a)openser.org > >To subscribe or unsubscribe via the World Wide Web, visit > http://openser.org/cgi-bin/mailman/listinfo/users >or, via email, send a message with subject or body 'help' to > users-request(a)openser.org > >You can reach the person managing the list at > users-owner(a)openser.org > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of Users digest..." > > >Today's Topics: > > 1. Re: deactivate an account (Helge Waastad) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Thu, 05 Jan 2006 11:23:30 +0100 >From: Helge Waastad <helge(a)smartnet.no> >Subject: Re: [Users] deactivate an account >To: maillisting(a)gmail.com >Cc: users(a)openser.org >Message-ID: <1136456610.11157.10.camel(a)localhost.localdomain> >Content-Type: text/plain > >Hi, >In the openserctl script there are acl options static configured >(free-pstn etc). >Just add inactive on the same line, and it should work > >br hw > > > > > > > > > > > >There is an error when I issue acl grant >openserctl acl grant myname(a)mydomain.com inactive >Invalid privilege: inactive ignored > > > >-- >Mvh/Br >Helge Waastad >Senior Engineer >Systemavdelingen >Smartnet > > > >------------------------------ > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > >End of Users Digest, Vol 8, Issue 11 >************************************ = = = = = = = = = = = = = = = = = = = = 　　　　　　　　致礼！　　　　　　　　Kenny Yeh 　　　　　　　　kenny(a)artdio.com.tw 　　　　　　　　　　2006-01-06

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