sr-users January 2007

sr-users@lists.kamailio.org

151 participants
232 discussions

by zahra bahar

Hi. Could any one say me when we use proxy server and when it is necessary to use redirect server? thanks. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

17 years, 3 months

[Users] TM : BYE retransmissions

by Jean-François SMIGIELSKI

Hello list! I observed, on an openser-1.1.0, that stateful relayings of BYE requests absorb retransmissions from the sender but do not retransmit it to the next hop in the Route header. Please, is it normal or is ti a misconfiguration from myself ? Thanks a lot! JF Smigielski. ________________________________________________________________________ iBELGIQUE, exprimez-vous ! http://web.ibelgique.com/

17 years, 3 months

Re: [Serusers] aliases

by Nils Ohlmeier

As this might be interesting for other users I cc'ed the list again. Just to be sure: you know that their is no alias table any more, right? Alias or not is just a matter of the flag value in the uri table. In general there is no need any more for re-writting a request which was targeted towards an alias. Request towards an alias are handled exactly the same way as "normal" requests, because both will get the same UID value. If you used the alias table for re-distributing the requests outside of your domain, which is actually more a forwarding function then an alias, then you could have this function as one (out of many) possible alternative(s) with AVPs like this: (Note: non-tested pseudo code) lookup_user("$t.uid", "ruri"); ... if ($t.uid) { load_attrs("$tu", "$t.uid"); if ($t.forwarding_target) { attr2uri("$t.forwarding_target"); t_relay(); } } Greetings Nils On Friday 05 January 2007 17:27, you wrote: > Thanks for the reply, Nils! I suppose I am a more of a newbie than I first > appeared as your answer is a little over my head. What I am trying to do > with the new version of SER is what used to be done (in previous versions) > with > > If (lookup("aliases")){ > Route(1); > > To redirect users based on a few entries in the aliases table. I won't be > using the SER server as a registrar or to handle dynamic user locations or > even authenticated accounts -- it will be open to the public without > authentication to access SIP accessible numbers and users. Mostly what I > want to do is just to route calls that SER receives (like john(a)iptn.org, > 5555555555(a)iptn.org etc.) to the domain/uri that used to be accessed in the > aliases table (openser worked this way too). > > Sorry for not explaining better and thanks again for your help! > > Eliott > > > -----Original Message----- > From: Nils Ohlmeier [mailto:nils@iptel.org] > Sent: Friday, January 05, 2007 8:32 AM > To: serusers(a)lists.iptel.org > Cc: Eliott Spencer > Subject: Re: [Serusers] aliases > > Hello, > > On Friday 05 January 2007 04:27, Eliott Spencer wrote: > > I am coming over from OpenSER because I ran some tests and found SER to > > be much faster. There are some things that are new to me that I imagine > > are easy to do. I have a working config with the 12/22/06 development > > snapshot > > > of 0.10 but I would like to lookup aliases to redirect inbound calls to > > another domain. I filled in the domain table and the uri table with a > > user > > > and I looked through the sample config but can't seem to get it to work. > > I > > > have also loaded the domain module, then the uri module, then the avp and > > avpops modules (among others). > > > > > > > > Any idea what I might need to put in my route? This doesn't seem to work > > (stolen from ser.cfg.sample): > > > > > > > > if (lookup_user("$t.uid", "@ruri")) { > > > > setflag(1); > > > > route(1); > > > > break; > > > > }; > > from looking at your code snipset I'm not sure what you really want to > achieve. > In general you should just do the lookup_user and then examine the AVPs > afterwards. I think that it is easier then looking at the result code of > the > > lookup_user function. > If you really want to distingiush if the call was targeted to an alias or > the > non-alias URI of an user then check for the AVP ruri_canonical. If it is > set > > (to the value 1) the URI contained the canonical (=non-alias) URI of the > user. > If you just want to know if the URI is a know one and belongs to a user, > just > check for the presence of the uid AVP after the lookup_user call. If it is > present the user is know to the system under the value of the uid AVP. > Otherwise the user/URI is not known to the system. > > Hope this helps > Nils

17 years, 3 months

[Users] forking call to another user on same proxy

by Christian Schlatter

I try to fork a call to another user registered at the same proxy as the caller. I figured out that the easiest method would be to use the append_branch() function, something like route[0] { if (method=="REGISTER") { save("location"); exit; } if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; } # if someone calls test2@domain, fork call to test4@domain # (test2 and test4 are both registered at domain) if (avp_check("$rU", "eq/test2")) { append_branch("sip:test4@127.0.0.1"); } if (!t_relay()) { sl_reply_error(); }; exit; } I know that this could also be achieved by adding permanent location records, but I'd like to load the forking targets from a DB in the future. So my first question would be if using append_branch("sip:test4@127.0.0.1"); is the correct way to do parallel forking to a local user. I'm asking because this method does not work for me as expected. It seems that the two UAC state machines running on the same proxy host that are involved in a forking call do somehow not work well together. Forking the INVITE to both test2 and test4 works well, but when I pick up test2 the call gets terminated after a few seconds. Further investigations have revealed that canceling the second call leg does not work correctly. There is a message flow diagram available at http://www.unc.edu/~cschlatt/openser.html . For some reason, the proxy process sending from 127.0.0.1 is retransmitting the 487 "Request Terminated" although it receives ACKs from the upstream proxy. And this causes the upstream proxy to send a 487 back to the caller (time 15.360), which then terminates the call established between caller and test2. And this only happens if I fork to the same proxy, if I use two proxy hosts everything works fine (??). Any advice is highly appreciated, Christian

17 years, 3 months

[Users] integrating with Asterisk and OpenSER for Voicemail

by raviprakash sunkara

Hi Users, I'm Setting UP the Voicemails by integrating with Asterisk and OpenSER, After 32 sec or 6 ring, it has to go the Voicemail server of Asterisk, In openser.cfg ........... is not hiiting the Asterisk server ............. ... any one help me ........ .... .... modparam("tm","fr_timer",6) modparam("tm","fr_inv_timer",24) modparam("tm","wt_timer",1) #mrodparam("tm", "ruri_matching", 0) #modparam("tm", "via1_matching", 0) modparam("avpops","avp_url","mysql://root:passwd@192.168.2.75/openser") modparam("avpops", "avp_table", "usr_preferences") modparam("avpops","avp_aliases","inv=i:15") ............... route { ...... if (loose_route()) { t_relay(); exit; }; if(is_method("INVITE") && !has_totag()) { xdbg("user [$ruri] has voicemail redirection enabled\n"); # backup R-URI avp_write("$ruri","$avp(inv)"); setflag(2); }; .............. .... route(1); } route[1] { if(isflagset(2)) { t_on_failure("1"); }; } failure_route[1] { log("----------------------------------------- \n"); if (t_was_cancelled()) { xdbg("transaction was cancelled by UAC\n"); return; } # restore initial uri avp_pushto("$ruri", "$avp(inv)"); prefix("9"); # route to Asterisk Media Server rewritehostport("192.168.2.75:5060"); append_branch(); t_relay("192.168.2.75:5060"); resetflag(2); } ......................... -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

17 years, 3 months

[Users] openser+freeradius:(radius_is_user_in(): Failure)+(DEBUG:avpops:ops_c)

by buptzhaoyuan＠sina.com

Hi, Lists,happy new year! Probably a stupid question,but I can not solved it correctly.The ua can not register the openser normally.I think the 5 step runs well and the last is down.The debug shows two errors. The first error is " 0(5898) radius_is_user_in(): Failure" ;the other is " 0(5898) DEBUG:avpops:ops_check_avp: no src avp found".So please give me help and some suggestion. And the fellowing is my configure file and debug infomation.Thanks!! HTTP/SIP RADIUS +-----+ (1) +-----+ +-----+ | |==========>| | | | | | (2) | | | | | |<==========| | | | | | (3) | | | | | |==========>| | | | | A | | B | (4) | C | | | | |---------->| | | | | | (5) | | | | | |<----------| | | | (6) | | | | | |<==========| | | | +-----+ +-----+ +-----+ ====> HTTP/SIP ----> RADIUS I setup the system by using the document(http://www.openser.org/docs/openser-radius-1.0.x.html) and my openser is openser-1.1.0-tls, the file openser.cfg is the same as the document. # #$Id$ # # radius config script # # ----------- global configuration parameters ------------------------ debug=7 # debug level (cmd line: -dddddddddd) fork=no 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 listen=udp:192.168.168.3 alias="swifton.org" alias="192.168.168.3" #fifo="/tmp/openser_fifo" # ------------------ module loading ---------------------------------- [root@localhost openser]# cat openser.cfg # #$Id$ # # radius config script # # ----------- global configuration parameters ------------------------ debug=7 # debug level (cmd line: -dddddddddd) fork=no 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 listen=udp:192.168.168.3 alias="swifton.org" alias="192.168.168.3" #fifo="/tmp/openser_fifo" # ------------------ module loading ---------------------------------- mpath="/usr/local/lib/openser/modules" loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "avpops.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "xlog.so" loadmodule "uri.so" loadmodule "acc.so" loadmodule "auth.so" loadmodule "auth_radius.so" loadmodule "group_radius.so" loadmodule "avp_radius.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc","db_url","mysql://openser:openserrw@localhost/openser") modparam("usrloc", "db_mode", 2) # -- acc params -- modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 1) modparam("acc", "service_type", 15) modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp") modparam("acc|auth_radius|group_radius|avp_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") # -- group_radius params -- modparam("group_radius", "use_domain", 1) # -- avpops params -- modparam("avpops", "avp_aliases", "day=i:101;time=i:102") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # check if user is suspended if(is_method("REGISTER|INVITE|MESSAGE|OPTIONS|SUBSCRIBE")) { if (radius_is_user_in("From", "suspended")) { sl_send_reply("403", "Forbidden - suspended"); exit; }; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); if(is_method("BYE")) { # log it all the time acc_rad_request("200 ok"); acc_log_request("200 ok"); } route(1); }; if(is_method("INVITE") && !has_totag()) { # set the acc flags setflag(1); setflag(2); }; if (!uri==myself) { # check if user is allowed to do voip calls to other domains if(is_method("INVITE|MESSAGE")) { if (!radius_is_user_in("From", "voip")) { sl_send_reply("403", "Forbidden VoIP"); exit; }; }; # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { # authenticate registers if (method=="REGISTER") { if (!radius_www_authorize("swifton.org")) { www_challenge("swifton.org", "0"); exit; }; # check the src ip address if(!avp_check("$avp(i:2)", "eq/$src_ip/ig")) { sl_send_reply("403", "Forbidden IP"); exit; }; save("location"); exit; }; # calls to pstn if(uri=~"sip:00[1-9][0-9]+@") { if(is_method("INVITE") && !has_totag()) { if (!radius_is_user_in("From", "pstn")) { sl_send_reply("403", "Forbidden PSTN"); exit; }; }; # set gateway address rewritehostport("10.10.10.10:5090"); route(1); }; # load callee's avps if(avp_load_radius("callee")) { # check if user has time filter enabled if(avp_check("$avp(i:3)", "eq/i:1")) { # print time in an avp avp_printf("$avp(i:100)", "$Tf"); # extract day avp_subst("$avp(i:100)/$avp(i:101)", "/(.{3}) .+/*\1*/"); if(!avp_check("$avp(i:6)", "fm/$day")) { sl_send_reply("403", "Forbidden - day"); exit; }; # extract 'hours:minutes' avp_subst("$avp(i:100)/$avp(i:102)", "/(.{10}) (.{5}):.+/\2/"); if((is_avp_set("$avp(i:4)") && avp_check("$avp(i:4)", "gt/$time")) || (is_avp_set("$avp(i:5)") && avp_check("$avp(i:5)", "lt/$time"))) { sl_send_reply("403", "Forbidden - time"); exit; }; }; }; ...... route(1); } ...... exit; } # and the openser start as follows: 898) SIP Request: 898) method: <REGISTER> 898) uri: <sip:swifton.org> 898) version: <SIP/2.0> 898) parse_headers: flags=2 898) end of header reached, state=5 898) parse_headers: Via found, flags=2 898) parse_headers: this is the first via 898) After parse_msg... 898) preparing to run routing scripts... 898) parse_headers: flags=100 898) DEBUG:maxfwd:is_maxfwd_present: value = 70 898) parse_headers: flags=10 898) DEBUG: add_param: tag=f59c388b5de348778e76ebda993c2d8e 898) DEBUG: add_param: epid=b6757bf3fc 898) DEBUG:parse_to:end of header reached, state=29 898) DBUG:parse_to: display={}, ruri={sip:zhaoy@swifton.org} 898) radius_is_user_in(): Failure 898) parse_headers: flags=200 898) DEBUG:parse_to:end of header reached, state=10 ...... 0(5898) grep_sock_info - checking if host==us: 11==13 && [swifton.org] == [192.168.168.3] 0(5898) grep_sock_info - checking if port 5060 matches port 5060 0(5898) grep_sock_info - checking if host==us: 11==13 && [swifton.org] == [192.168.168.3] 0(5898) grep_sock_info - checking if port 5060 matches port 5060 0(5898) check_nonce(): comparing [459bb612ad569015987d4849d1ae603515f32e6b] and [459bb612ad569015987d4849d1ae603515f32e6b] 0(5898) DEBUG:auth_radius:radius_authorize_sterman: Success 0(5898) DEBUG:auth_radius:generate_avps: getting SIP AVPs from avpair 225 0(5898) DEBUG:auth_radius:extract_avp: string is <Sip-Group:voip> 0(5898) DEBUG:auth_radius:extract_avp: AVP name is <Sip-Group> 0(5898) DEBUG:auth_radius:extract_avp: AVP val is <voip> 0(5898) DEBUG:auth_radius:generate_avps: AVP 'Sip-Group'/0='voip'/0 has been added 0(5898) DEBUG:avpops:ops_check_avp: no src avp found 0(5898) parse_headers: flags=ffffffffffffffff 0(5898) check_via_address(192.168.168.30, 192.168.168.30, 0) 0(5898) DEBUG:destroy_avp_list: destroying list 0x422bc440 0(5898) receive_msg: cleaning up the freeradius debug infomation as follows: er-Name = "zhaoy(a)swifton.org" Digest-Attributes = 0x0a077a68616f79 Digest-Attributes = 0x010d73776966746f6e2e6f7267 Digest-Attributes = 0x022a34353962623631326164353639303135393837643438343964316165363033353135663332653662 Digest-Attributes = 0x04117369703a73776966746f6e2e6f7267 Digest-Attributes = 0x030a5245474953544552 Digest-Response = "2ead4f5b9d78013a35f2eb7a792971a5" Service-Type = Sip-Session Sip-Uri-User = "zhaoy" NAS-Port = 5060 NAS-IP-Address = 192.168.168.3 Processing the authorize section of radiusd.confmodcall: entering group authorize for request 167 modcall[authorize]: module "preprocess" returns ok for request 167 modcall[authorize]: module "chap" returns noop for request 167 modcall[authorize]: module "mschap" returns noop for request 167 rlm_digest: Converting Digest-Attributes to something sane... Digest-User-Name = "zhaoy" Digest-Realm = "swifton.org" Digest-Nonce = "459bb612ad569015987d4849d1ae603515f32e6b" Digest-URI = "sip:swifton.org" Digest-Method = "REGISTER"rlm_digest: Adding Auth-Type = DIGEST modcall[authorize]: module "digest" returns ok for request 167 ....... Processing the authenticate section of radiusd.confmodcall: entering group authenticate for request 167A1 = zhaoy:swifton.org:zhaoyA2 = REGISTER:sip:swifton.orgH(A1) = eff77a9105c5c0973492694f59c944f2H(A2) = d3add39dad2709bea55016ba79eb675aKD = eff77a9105c5c0973492694f59c944f2:459bb612ad569015987d4849d1ae603515f32e6b:d3add39dad2709bea55016ba79eb675aEXPECTED 2ead4f5b9d78013a35f2eb7a792971a5RECEIVED 2ead4f5b9d78013a35f2eb7a792971a5 modcall[authenticate]: module "digest" returns ok for request 167modcall: group authenticate returns ok for request 167radius_xlat: 'Authenticated'Login OK: [zhaoy(a)swifton.org] (from client openser port 5060)Sending Access-Accept of id 13 to 192.168.168.3:33101 SIP-AVP = "Sip-Group:voip" Reply-Message = "Authenticated" Sip-Group = "voip"Finished request 167 and my MSN:buptzhaoyuan@sina.com .I am nice to communicate with you! ------------------------------------------------------------------- 惠普商用台式机dc5750,高性能，低功耗！( http://ad4.sina.com.cn/sina/limeng3/mail_zhuiyu/2006/mail_zhuiyu_20061225.h… ) =================================================================== 注册新浪2G免费邮箱（ http://mail.sina.com.cn/chooseMode.html ）

17 years, 3 months

[Users] Reg:Instance messages

by venkid

hai all, i am using X-lite as sip client .i have installed in window and am able to send instance messages and presence info is also fine with openser In redhat Linux i am trying to send instance messages using X-lite , but i am not able send and not able to receive any messages . is X-lite linux version is support for redhat linux Es 3.0 or any other sip client is there for redhat linux Es 3.0 Thanks and Regards venkatesh.d

17 years, 3 months

[Serusers] aliases

by Eliott Spencer

I am coming over from OpenSER because I ran some tests and found SER to be much faster. There are some things that are new to me that I imagine are easy to do. I have a working config with the 12/22/06 development snapshot of 0.10 but I would like to lookup aliases to redirect inbound calls to another domain. I filled in the domain table and the uri table with a user and I looked through the sample config but can't seem to get it to work. I have also loaded the domain module, then the uri module, then the avp and avpops modules (among others). Any idea what I might need to put in my route? This doesn't seem to work (stolen from ser.cfg.sample): if (lookup_user("$t.uid", "@ruri")) { setflag(1); route(1); break; }; Thanks!

17 years, 3 months

[Serusers] PA module segfault

by TERRY ROBERT

Hi, I'm getting a segfault error while running the latest presence server snapshot (ser-0.10.99-dev35-pa-4.2). The scenario is: UA SER SUBSCRIBE ----------> 200 <---------- NOTIFY <---------- 200 ----------> Everything is fine until the subscription expires, at which point the PA module segfaults and generates a core file. Any suggestions on this? (logfile and backtrace from gdb are attached).

17 years, 3 months

Re: [Users] LCR db_only mode problem

by Jayesh Nambiar

Hi Juha, Thanks for your reply. First prominent doubt that I have is; when I am using LCR in db_only mode why does openser try to load the table while starting up? The scenario is as follows: I had 50 rows in LCR table and started openser and it started properly and also routed the calls properly according to the values in lcr and gw table. Now I just appended more 40,000 rows to this table and again the routing worked properly as per my lcr table. This behaviour was correct as I was using query only mode. I did not need any restart or any fifo reload command. Only when I restarted my openser, the openser again died as it was trying load the lcr table while starting. Could this be a bug in lcr module as I am not sure whats happening. I tried increasing MAX_NO_OF_LCRS but even that did not help. Somebody please help if poosible. Thanks, w/regards, jayesh ----- Original Message ---- From: Juha Heinanen <jh(a)tutpro.com> To: Jayesh Nambiar <voip_freak(a)yahoo.co.in> Cc: openser <users(a)openser.org> Sent: Tuesday, 26 December, 2006 10:58:01 PM Subject: [Users] LCR db_only mode problem X-VM-v5-Data: ([nil nil nil nil nil nil nil nil nil] the error messages you get are coming from this statement: if (lcr_dbf.query(dbh, NULL, 0, NULL, lcr_cols, 0, 4, 0, &res) < 0) { LOG(L_ERR, "lcr_reload_gws(): Failed to query lcr data\n"); lcr_dbf.close(dbh); return -1; } i don't know which kind of memory db queries reserve. if shared memory, you can start openser with larger -m parameter value. if pkg memory, you need to edit PKG_MEM_POOL_SIZE in config.h and recompile openser. with lots of lcr entries and no caching, performance may be bad. if you use caching then you need to increase value of #define MAX_NO_OF_LCRS 256 in lcr_mod.c and recompile lcr module. increasing MAX_NO_OF_LCRS means that more shared memory will be used so you may also need to start openser with larger -m parameter value. -- juha Send free SMS to your Friends on Mobile from your Yahoo! Messenger. Download Now! http://messenger.yahoo.com/download.php

17 years, 3 months

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sr-users January 2007