- sr-users - kamailio.org

[OpenSER-Users] newbie help - handeling 302s and 404s
by Lazer Ramm 18 Dec '07

18 Dec '07

Hi, i need a simple config that achieves the following (w/ out a db) if a user is not registered then all calls should go to a default gw 192.168.10.1 the default gw will only respond w/ a 302 or a 404 if a 302 is received then openser should forward the call on to the new gw listed in the reply if a 404 is received the call should be forwarded to a different gw 172.16.20.1 if the send gw is done then the call should die. here is what i was thinking of but as a newbie i may be off my rocker i seem to get an error ERROR: t_check_status: cannot check status for a reply which has no T-state established but again my whole approach may be wrong. if (!lookup("location")) { rewritehostport ("192.168.10.1:5060"); if (t_check_status("(404)|(480)")) { log("404 or 480 negative reply\n"); rewritehostport ("172.16.20.1:5060"); } else{ t_on_failure("1"); t_relay(); } failure_route[1] { get_redirects("*"); t_relay(); } } thanx in advance

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Re: [OpenSER-Users] Want to know about configuring openser as strict router
by Vikas CG. 18 Dec '07

18 Dec '07

Hi, Thanks for the information, Actually we were looking to some server which can acts as a strict router. Can u suggest regarding this? Thanks, Vikas C.G. -----Original Message----- From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Tuesday, December 18, 2007 7:59 PM To: Vikas CG. Cc: users(a)openser.org Subject: Re: [OpenSER-Users] Want to know about configuring openser as strict router Hi! Openser is a loose-router. This means that the inserted Record-Route header do have the ;lr parameter. Thus, other SIP clients know that openser is a looserouter and weill send messages in loose-router syntax. On the other hand, openser checks if the next route header and/or the RURI contains the ;lr parameter to make conversion from/to strict-router syntax. Thus, depending on if the previous and next hops are strict or loose routers, openser will do the conversion (by using the loose_route() function). regards klaus Vikas CG. schrieb: > Hi all, > > > > We are using openser server for testing. Can we configure it as > strict router? > > I have tried making changes in configuration file by enabling Record > Route. > I want to know whether the openser will strictly acts as "Strict Router" > I.e. performing actions like modifying Request URI and last route header. > Can any body suggest me? > > > > Thanks in advance , > Vikas C.G. > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users

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[OpenSER-Users] OpenSER: AVP script flag bugs in 1.3 and trunk
by Helmut Kuper 18 Dec '07

18 Dec '07

Hello, in openser 1.3 and current trunk the AVP script flags still not work as expected. So here again the fixes I did to solve that problems in 1.3 and trunk: ... Hi again, further tests with avp and script_flags raised new problems. When I write resp. set an AVP like "$avp(i1:10)=10; $avp(i2:10)=20;" and checking them with "xlog('L_INFO', 'avp(i1:10) avp(i1:20)\n');" it leads to "20 20" as output. In file pvar.c in function pv_get_avp_name() I had to replace *name_type = 0; with *name_type = ip->pvn.u.isname.type; Then that worked. - -.- Additionally I fixed the Alias-Duplicate Problem when you do something like avp_aliases("A=i1:10; B=i2:20"); In file usr_avp.c I had to change line 427 in function check_avp_galias() from type &= AVP_NAME_STR; to type &= (AVP_NAME_STR|AVP_SCRIPT_MASK); - -.- Then at last, if u want assign a value to an AVP-Alias which uses script flags replace in file usr_avp.c in function add_avp_galias() around line 482 the line ga->avp.type = type&AVP_NAME_STR; with ga->avp.type = type&(AVP_NAME_STR|AVP_SCRIPT_MASK); regards Helmut

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[OpenSER-Users] Want to know about configuring openser as strict router
by Vikas CG. 18 Dec '07

18 Dec '07

Hi all, We are using openser server for testing. Can we configure it as strict router? I have tried making changes in configuration file by enabling Record Route. I want to know whether the openser will strictly acts as "Strict Router" I.e. performing actions like modifying Request URI and last route header. Can any body suggest me? Thanks in advance , Vikas C.G.

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[OpenSER-Users] Question about feature requests
by Sergio Gutierrez 18 Dec '07

18 Dec '07

Hello everybody. I would like to know how is the right procedure to implement a new feature for OpenSER. For example, if I propose a new Feature, is it well to include the patch for the implementation, or I just propose it, and wait for the approval of one of the core developers? Best regards. Sergio Gutiérrez.

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[OpenSER-Users] ERROR:core:io_watch_add: epoll_ctl failed: Invalid argument [22]
by Mihail Gluhowchenko 18 Dec '07

18 Dec '07

Truing to start openser-1.3.0. In log i`m saw: ERROR:core:io_watch_add: epoll_ctl failed: Invalid argument [22] Dec 18 12:37:17 devel s-eltex[27564]: CRITICAL:core:tcp_main_loop: failed to add process 5 (TCP main) unix socket to the fd list Dec 18 12:37:17 devel s-eltex[27564]: CRITICAL:core:tcp_main_loop: exiting... In config i`m set listen=udp:192.168.138.59:5060 Why openser truing to open tcp sockets? And another bug: If stop openser and then start. If dont switch off SIP clients server down with error.

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[Serusers] ser multi-domain support SOLVED
by satish patel 18 Dec '07

18 Dec '07

Dear all I have solve my multidomain configuration with ser-0.9.6 and it is working fine for me and i am sending you my configuration for incomming users whos want do some stuff wid ser i have find many document on multi-support and i have consolidate all document and make my ser.cfg script or config cat /usr/local/etc/ser/ser.cfg # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=7 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=192.168.104.20 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/domain.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/call.so" #loadmodule "/usr/local/lib/ser/modules/vm.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", 0) # -- 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("domain", "db_url", "mysql://root:mypasswd@localhost/ser") modparam("domain", "db_mode", 1) # Use chaching modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") modparam("usrloc|registrar|auth_db|avpops|group", "use_domain", 1) #modparam("usrloc", "use_domain", 1) #modparam("registrar", "use_domain", 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 > max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); 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) #-----------Multi-domain Logic work blow in stanza--------# if (is_from_local()) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("", "subscriber")) { www_challenge("", "0"); break; }; save("location"); break; save("aliases"); break; }; #----Alias check--- # lookup("aliases"); # if (!uri==myself) { # route(1); # break; # }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } #-----END---# Now to test configuration start ser #serctl start #serctl domain add abc.com #serctl add 222(a)abc.com tulip 222(a)abc.com #serctl add 333(a)abc.com tulip 333(a)abc.com this to user 222 and 333 for domain abc.com **Notes:- you should configure SRV recode on your DNS otherwise it will not work for you NOW configure your Xlite and try to call each other Then try to add new domain and do same thing which u have done before abc.com Thanks and Best of Luck $ cat ~/satish/url.txt http://www.linuxbug.org _____________________________________________________________________________________________________ --------------------------------- Share files, take polls, and discuss your passions - all under one roof. Click here.

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[Serusers] call.so and vm module missing
by satish patel 18 Dec '07

18 Dec '07

Dear all I have compileing ser-0.9.6 and it is compiled successful but when i going to /usr/local/lib/ser/module directory there is no required module call.so and vm.so how to compile i have also try this one make modules=modules/call modules but there is no file found How to compile and get support of call.so module and vm.so module $ cat ~/satish/url.txt http://www.linuxbug.org _____________________________________________________________________________________________________ --------------------------------- Get the freedom to save as many mails as you wish. Click here to know how.

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[Serusers] Ser sends messages not in that order in which it has received
by Alexandr Volkov 18 Dec '07

18 Dec '07

Hi, We have found one unpleasant situation with a Ser often shown at greater loadings. At greater loadings and at work of a Ser with number of child processes greater than one (Parameter 'children' in ser.cfg), Ser sends messages not in that order in which it has received. And as the result appears request pending errors. We simulated this situation continuously reinviting the remote UA. Usually the error is occured on the tenth-twentieth reinvite. In the attachment there are SIP logs reproducing this situation. Somebody can tell in occasion of this problem, or offer the decision? Or whether there is a version in which there is no this lack? Alexandr

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[OpenSER-Users] OpenSER - unixODBC Module
by Martin.Gruber1＠infineon.com 18 Dec '07

18 Dec '07

Hi, Well I now know my problem why I cannot connect to my Oracle-Database. The Connectionport hast changed. My question is, how can I insert the Port-Number the the connectionstring in openser.cfg....... unixodbc://user:password@oracle.server.com:PortNr/ORA Does not work at all. Does anybody know, where I can add the Port Number in the Connectionstring? Kind regards, Martin

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Re: [OpenSER-Users] OpenXcap exception: TypeError in twisted.web2.stream.IByteStream
by Antoine Fressancourt 18 Dec '07

18 Dec '07

Hello, I have the same error as Gregorio on my installation, I am running OpenSER 1.3 and my XCAP client is eyeBeam². Thanks in advance for your help, Antoine 2007/10/30, Gregorio <goyo.villalba(a)dialcom.com>: > > Hi all. > > > > I get an exception in openXcap when I send a GET to retrieve the resources > file. > > I explain the process I followed: > > > > ----->Send GET to openXcap. > > <----Receive 404, because there is not file. > > > > -----> Send PUT to openXcap. The body contains a resource-list.xmldocument 'with no data': > > Body; > > <?xml version='1.0' encoding='UTF-8'?> > > <resource-lists xmlns='urn:ietf:params:xml:ns:resource-lists' > xmlns:cp='urn:ietf:params:xml:ns:copycontrol'> > > <list></list> > > </resource-lists> > > > > <----Receive 200 OK. So openXcap likes my document. > > > > -----> Send GET to openXcap. > > <---- Receive 500 "An error occurred rendering the requested page. More > information is available in the server log". So the same document that was > accepted in the PUT throws an exception. > > > > The server log says: > > > > Oct 30 15:01:55 localhost openxcap[2791]: [-] Exception rendering: > > Oct 30 15:01:55 localhost openxcap[2791]: [-] Traceback (most recent call > last): > > Oct 30 15:01:55 localhost openxcap[2791]: [-] File > "/usr/lib/python2.4/site-packages/twisted/internet/posixbase.py", line 228, > in mainLoop > > Oct 30 15:01:55 localhost openxcap[2791]: [-] self.runUntilCurrent > () > > Oct 30 15:01:55 localhost openxcap[2791]: [-] File > "/usr/lib/python2.4/site-packages/twisted/internet/base.py", line 533, in > runUntilCurrent > > Oct 30 15:01:55 localhost openxcap[2791]: [-] f(*a, **kw) > > Oct 30 15:01:55 localhost openxcap[2791]: [-] File > "/usr/lib/python2.4/site-packages/twisted/internet/defer.py", line 239, in > callback > > Oct 30 15:01:55 localhost openxcap[2791]: [-] > self._startRunCallbacks(result) > > Oct 30 15:01:55 localhost openxcap[2791]: [-] File > "/usr/lib/python2.4/site-packages/twisted/internet/defer.py", line 304, in > _startRunCallbacks > > Oct 30 15:01:55 localhost openxcap[2791]: [-] self._runCallbacks() > > Oct 30 15:01:55 localhost openxcap[2791]: [-] --- <exception caught > here> --- > > Oct 30 15:01:55 localhost openxcap[2791]: [-] File > "/usr/lib/python2.4/site-packages/twisted/internet/defer.py", line 317, in > _runCallbacks > > Oct 30 15:01:55 localhost openxcap[2791]: [-] self.result = > callback(self.result, *args, **kw) > > Oct 30 15:01:55 localhost openxcap[2791]: [-] File > "/usr/lib/python2.4/site-packages/xcap/resource.py", line 51, in > sendResponse > > Oct 30 15:01:55 localhost openxcap[2791]: [-] response = > http.Response(response.code, stream=response.data) > > Oct 30 15:01:55 localhost openxcap[2791]: [-] File > "/usr/lib/python2.4/site-packages/twisted/web2/http.py", line 109, in > __init__ > > Oct 30 15:01:55 localhost openxcap[2791]: [-] self.stream = > IByteStream(stream) > > Oct 30 15:01:55 localhost openxcap[2791]: [-] exceptions.TypeError: > ('Could not adapt', array('c', "<?xml version='1.0' > encoding='UTF-8'?><resource-lists > xmlns='urn:ietf:params:xml:ns:resource-lists' > xmlns:cp='urn:ietf:params:xml:ns:copycontrol'><list></list></resource-lists>"), > <InterfaceClass twisted.web2.stream.IByteStream>) > > > > > > Anyone know what is wrong? > > Thanks and best regards. > > Gregorio > > > > > > > > > > > > > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users > >

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[OpenSER-Users] publish with empty body
by jh＠tutpro.com 18 Dec '07

18 Dec '07

while testing a new sip ua, i noticed that at some point it sent (don't ask me why because i don't know) a publish request without any body. it was not an initial publication, because it contained SIP-If-Match header with an etag. the publication was for even package presence. what openser presence server did was that it accepted the publish with 200 ok that contained a NEW etag value in its SIP-If-Match header. i have been trying to find out from rfcs where this kind of behavior would have been defined, but so far without success. rfc3903 says: 5. The ESC processes the published event state contained in the body of the PUBLISH request. If the content type of the request does not match the event package, or is not understood by the ESC, the ESC MUST reject the request with an appropriate response, such as 415 (Unsupported Media Type), and skip the remainder of the steps. in this publish content type didn't match the event package, because there was none. reading further i found this: * If the request has no message body and contained no entity-tag, the ESC SHOULD reject the request with an appropriate response, such as 400 (Invalid Request), and skip the remainder of the steps. Alternatively, in case either ESC local policy or the event package has defined semantics for an initial publication containing no message body, the ESC MAY accept it. but it does not apply, because the publication was not initial. so, have i found a bug or any other ideas? -- juha

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Re: [Serusers] rewritehostport and RFC3261 compliant
by Klaus Darilion 18 Dec '07

18 Dec '07

Hi Adrien! I Cc this email also to the seruser mailing list as it might be interesting for others too. Adrien Lemoine schrieb: > Hi Klaus, > > I've some good news !!! (Sorry to take your time) > > I try the solution in the URL paste in the last mail, and it runs ! > > So I've something like this : > > if ( t_check_status("(1[0-7][0-9])|(180)") ) { > xlog("L_WARN","time=[%Tf] call id=[%ci] call seq=[%cs] contact > header=[%ct] from uri=[%fu] from tag=[%ft] request's method=[%rm] request's > uri=[%ru] to uri=[%tu] to tag=[%tt] sip message id=[%mi] process id=[%pp] ip > source=[%is] flags=[%mf], fix To-Tag failure [Thomson]\n"); > subst('/^(To:.*)\;tag=[^;[:cntrl:]]*/\1/i'); > } > break; > > The guy precise that 183 don't have to be included in this test, why ? Bogdan was afraid that this could break "early media", but I'm sure it is best to include 183 as well. Thus this regular expression should handle all provisional responses. t_check_status("(1[0-9][0-9])") > Is this could be okay for simplify the solution and don't take the risk to > break an other service by collateral effects ? You never know and it would be better to fix it in the SIP phone/gateway. But I think it is a acceptable workaround. > I think I beging to understand the problem : my Thomson product "remember" > of the first to-tag include in "180 ringing" and doesn't accept an other > to-tag for the same transaction. Isn't it ? Yes. Some theory: SIP has to match incoming requests and responses to existing dialog. This is done by comparing the 3-tuple call-id/from-tag/to-tag. Messages which are sent out of dialog do not have a to-tag (tag parameter in the To: header). The initial INVITE request is sent "out-of-dialog" (no to-tag). If a SIP client responses to the INVITE request it adds a to-tag to the response. Thus, if the INVITE is sent to multiple devices (e.g. parallel ringing on multiple SIP devices or call forwarding on timeout) the caller will receive multiple responses. As the to-tag is different for each called SIP device the caller can detect responses from different devices. (this is a feature of SIP). If the caller SIP client receives provisional responses (101-199) it will create an "early dialog". Once a final response is received the dialog will convert into "confirmed" state ( 200 OK) or the dialog will be terminated (3xx-6xx responses). Example: caller proxy phone1 phone2 *0 --INVITE----------> <-100 (no to tag)-- ----INVITE------> *1 <---100 trying--- <---180 ringing-- <---180 ringing----- *2 ----INVITE-----------------> <---100 trying-------------- <---180 ringing------------- <---180 ringing----- *3 <---200 ok----------------- <----200 ok --------- *4 ----CANCEL------> ........ An RFC compliant SIP client should handle the scenario following: *0: the caller sends the INVITE (out-of-dilaog). *1: The caller receives a 100 trying response. A "100 trying" does not have a to-tag and thus does not create dialogs. It will only stop retransmissions. *2: The caller receives a provisional response. It creates an "early-dialog" with the identifier call-id/fromtag/totag1. *3: The caller receives another provisional response. The response has a different to-tag. Thus the caller creates a second "early-dialog" with identifier call-id/fromtag/totag2. *4: The caller receives a final response. The 200 ok matches call-id/fromtag/totag2. Thus, the first early-dialog will be terminated and the second early-dialog will be transformed into "confirmed" state. Some non-complient clients handle it in other ways, e.g.: - Asterisk with "pedantic=no": Asterisk does not care about totags in provisional responses but only in the 200 ok. (works but insecure). - Asterisk with "pedantic=yes": Asterisk learns the totag from the first provisional response. All further responses with different totag are ignored. (breaks compatibility). I guess the Thomson device has the some bug as Asterisk in pedantic mode. > If it's that, I don't read something about this behaviour in the RFC to > argument my quick fix. Of course the workaround with "removing totags in provisional responses" breaks the RFC - but in the case with broken clients the RFC is already broken by the clients. > An other problem appears, you said that "try to remove the to-tag in the 180 > and 183 responses *to this client*". I would like too, but how can I match > UserAgent in Status, there's not. If the INVITE has a user agent header like "User-Agent: Thomsen abcd....." you activate the reply_route. E.g. ... if ((method == INVITE) && (search("User-Agent: Thomson")) { t_on_reply("1"); xlog("L_INFO","buggy Thomson, activate reply route to strip totag"); } ... onreply_route[1] { xlog("L_INFO","reply route: strip totag"); # put some more xlog here if wanted... if (t_check_status("(1[0-9][0-9])")) { subst('/^(To:.*)\;tag=[^;[:cntrl:]]*/\1/i'); } } regards Klaus

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[OpenSER-Users] Using append_hf("Route: xx") and loose_route() not stripping pre-loaded routeset?
by Taisto Qvist 18 Dec '07

18 Dec '07

Hi Guys, I have been trying all day, to create "static routing" in my openser-proxy(1.3). My openser-script is based on the default, so what I am doing is just: ------------------------------------------- if ( loose_route() ) { // strip of preloaded route is NOT being done here?? route(); } if ( isToMe() ) { // Do local logic } else { // forward this via proxyX append_hf("Route: <sip:1.1.1.1:5060;lr>\r\n"); if (loose_route() ) { route(1); } else { // I always end up here...loose_route() always returns false... // because its looking at the first route, that should have been stripped off... } ----------------------------- I have added lots of traces to lookup.c, and according to those, it seems that the preloaded route is not stripped off properly? Once I get too the second "loose_route()", that function IS STILL looking at the first preloaded route, which should have been stripped in the first "if". I am getting the feeling that the "deletion/stripping" of the first, preloaded, Route, is not done directly, causing the second check, to find the first... My work-around is naturally to use setdsturi(), but I thought that loose_route() should be able to do all this for me...? Also, this used to work, in version 1.1, about year ago :-) Any toughts? Regards Taisto Qvist

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[Serusers] SIP Express Bundle - let's get started
by Greger V. Teigre 18 Dec '07

18 Dec '07

Hi guys, thanks a bunch for lots of input and I really appreciate the willingness to contribute. Thus, I have created a "project page": http://www.iptel.org/sip_express_bundle_sip_service_in_15_minutes I thought we could gather the current perspective on that page and document our decisions as we go. It will hopefully be useful in our process, as well as documentation for us and others later. I have noted the following volunteers: Jai: testing and installation work ram: testing SIP: tesing Jiri: anyting? Mike: testing and documentation Maybe we should set up a small mailing list for coordination emails, but for now, let's use serusers (where I think all the comments came). (I copy the other lists on this post, so the other lists know that the discussion will move to serusers). Out of the comments, I read CentOS and vmware as the most wanted combination. I have documented the pros and cons on the project page, and suggest that we do some testing before we decide. I also tried out rpath (which I have no previous experience with). My observations are documented on the same page. Thus, I have started setting up a minimal CentOS virtual appliance found in vmware's appliance directory on an esx server. I will send details on accessing it to the volunteers once it is up and running (decompressing, unpacking, and building a non-split disk takes an awful lot of time :-( ). Ok, further comments, ideas, etc, please post to serusers or edit/add comments to the project page (requires an iptel.org account). I see the following steps with documentation as we go (steps also found on the project page): 1. Testing and specification of what we want to accomplish 2. Environment and OS setup to ensure that we easily can release new versions 3. Installation and configuration of the software. I assume this step also will involve development of some tools we need, as well as adaptation of existing stuff 4. Testing and user documentation 5. Packaging and deployment Let's get going! g-)

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[OpenSER-Users] invite resend bug?
by yoshiki 18 Dec '07

18 Dec '07

hi, I am usingo openser-1.2.0 When a client transmitted INVITE and transmitted INVITE once again because a reply of openser was slow, these two INVITE message was treated as another message. The peer transmitted 200OK for INVITE of the beginning, but a server transmits "404 not found" to next INVITE. Will even a message of the same call ID be handled in this way? yokkie -------------------------------------- New Design Yahoo! JAPAN 2008/01/01 http://pr.mail.yahoo.co.jp/newdesign/

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[OpenSER-Users] Preparation for 1.2.3 release
by Bogdan-Andrei Iancu 17 Dec '07

17 Dec '07

Hi, I would like to try to get 1.2.3 out before Christmas, so here is a starting plan: 1) do some backporting related to bug fixes (like in TM, with Route for local generated requests) 2) do some backporting related to compiling errors and warnings (solaris, 64 bits, BSD, etc) 3) backport the new default configuration script we have in 1.3.0 I can take care of all this, not a problem. Is anybody seeing any stopper for doing the release by Thursday 20? Regards, Bogdan

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[OpenSER-Users] OpenSER 1.3.0 release - updates
by Bogdan-Andrei Iancu 17 Dec '07

17 Dec '07

Hi, The sources are no available for download from the SF site also. See: https://sourceforge.net/project/showfiles.php?group_id=139143 In some case SF will be better to use (for this purpose) as they mirror the files in multiple locations. Also, I generated and uploaded on openser.org packages for debian unstable and testing. See: http://www.openser.org/pub/openser/latest/packages/debian-unstable/ During next week, packages for debian stable will be generated and uploaded on the project local apt repository. Also some other binary tarballs are promised (by other contributors) for different OS and archs. Regards, Bogdan

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[OpenSER-Users] FIFO / SEMS
by Bruno Machado 17 Dec '07

17 Dec '07

Hi all I am reading about FIFO and I am wandering in my thoughts, searching for a solution that tear down a session. I don't want to use Asterisk anymore for this purpose. Can I send a BYE message through FIFO? Should I use "t_uac_dlg"? I was reading about "t_uac_cancel" but it generates a CANCEL and, of course, it won't resolve the problem. I didn't install SEMS yet. So, I would like to know if someone uses this software (OpenSER + SEMS) within this purpose. This link: http://svn.berlios.de/wsvn/sems/trunk/doc/Readme.sw_prepaid_sip?op=file&rev… has this: "When the call duration exceeds the credit, the call is terminated by SEMS." So, I can realize it sends a BYE to OpenSER through FIFO and it will be forwarded to the voip devices. Am I right? I will try to use the 't_uac_dlg' now. Any comments are welcome. Cheers Bruno Machado Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento! http://br.mail.yahoo.com/

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Re: [OpenSER-Users] OpenXcap fails to start the HTTP on port
by Amit Vijayvargiya 17 Dec '07

17 Dec '07

Hi all , I have a problem in running openxcap server this is my syslog Starting Open XCAP 0.9.7 Dec 17 21:02:03 ankith openxcap[31833]: [-] Traceback (most recent call last): Dec 17 21:02:03 ankith openxcap[31833]: [-] File "/usr/bin/openxcap", line 56, in ? Dec 17 21:02:03 ankith openxcap[31833]: [-] from xcap.server import XCAPServer Dec 17 21:02:03 ankith openxcap[31833]: [-] File "/usr/lib/python2.3/site-packages/xcap/server.py", line 16, in ? Dec 17 21:02:03 ankith openxcap[31833]: [-] from twisted.web2 import channel, resource, http, responsecode, server Dec 17 21:02:03 ankith openxcap[31833]: [-] File "/usr/lib/python2.3/site-packages/twisted/web2/channel/__init__.py", line 7, in ? Dec 17 21:02:03 ankith openxcap[31833]: [-] from twisted.web2.channel.cgi import startCGI Dec 17 21:02:03 ankith openxcap[31833]: [-] File "/usr/lib/python2.3/site-packages/twisted/web2/channel/cgi.py", line 8, in ? Dec 17 21:02:03 ankith openxcap[31833]: [-] from twisted.web2 import http, http_headers, server, responsecode Dec 17 21:02:03 ankith openxcap[31833]: [-] File "/usr/lib/python2.3/site-packages/twisted/web2/http.py", line 27, in ? Dec 17 21:02:03 ankith openxcap[31833]: [-] from twisted.web2 import iweb Dec 17 21:02:03 ankith openxcap[31833]: [-] File "/usr/lib/python2.3/site-packages/twisted/web2/iweb.py", line 58, in ? Dec 17 21:02:03 ankith openxcap[31833]: [-] class IOldNevowResource (components.Interface): Dec 17 21:02:03 ankith openxcap[31833]: [-] AttributeError: 'module' object has no attribute 'Interface' I think problem in somewhere in zope interface ..............but I have install zope 3.3.0 version ......... please help me out ............ Regards Amit On Wed, 2007-10-17 at 13:11 +0200, Gregorio wrote: > Hi all. > > > > Finally (buff!!) I could configure openXcap, after executing ‘openxcap > start’ the log says: > > > > ‘ [-] Log opened. > > [-] Starting Open XCAP 0.9.4 > > [-] Supported Root URIs: http://192.168.1.206 > > [-] twisted.web2.channel.http.HTTPFactory starting on 8000’ > > > > And nothing else, no more logs displayed. OpenXcap process is running > in the machine: > > [root@XXXXX]# ps -ef|grep xcap > > root 926 1 0 18:02 ? > 00:00:00 /usr/bin/python /etc/rc.d/init.d/openxcap start > > > > But no one is listening in port 8000, neither before executing xcap > nor after. > > > > I haven´t found any error in any log so I don’t know what is going on. > > Anyone can help me? > > > > Thanks. > > Gregorio > > > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users

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Re: [Serusers] SER 2.0 installation - SER does not start due to missing modules
by Strassburger, Patrick 17 Dec '07

17 Dec '07

Hello Greger, I missed the mysql-devel package. Nevertheless now it claims on folowing: : [root@ser ser]# ser -c 0(9562) ERROR: load_module: could not open module </usr/local/lib/ser/modules/x mlrpc.so>: /usr/local/lib/ser/modules/xmlrpc.so: cannot open shared object file: No such file or directory 0(9562) parse error (75,13-14): failed to load module 0(9562) set_mod_param_regex: No module matching <xmlrpc> found 0(9562) parse error (142,20-21): Can't set module parameter 0(9562) parse error (235,16-17): unknown command, missing loadmodule? I installed php-xmlrpc-5.1.6-5.el5.i386.rpm and did another installation but still the same. (I did the following command for modules: "make group_include="standard mysql" all", "make group_include="standard mysql" installall") Do I miss anything here? Kind regards Patrick -----Ursprüngliche Nachricht----- Von: Greger V. Teigre [mailto:greger@teigre.com] Gesendet: Montag, 17. Dezember 2007 15:12 An: Strassburger, Patrick Cc: serusers(a)lists.iptel.org Betreff: Re: [Serusers] SER 2.0 installation - SER does not start due to missing modules You must do make install with the same group_include. Verify that you have mysql-devel and mysql rpm packages installed (required to compile and run). g-) Strassburger, Patrick wrote: Hello all, I am a newbie in SER and am failing since a few days installing SER 2.0. After several installation attempts I would like to ask you for some help. I did a "make group_include="standard mysql" all" but cannot see module mysql - where do I find those two modules and can I add them afterwards? Kind regards Patrick I get the following error and do not know how to resolve this: [root@ser ser]# ser -c 0(5613) ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: /usr/local/lib/ser/modules/mysql.so: cannot open shared object file: No such file or directory 0(5613) parse error (55,13-14): failed to load module 0(5613) ERROR: load_module: could not open module </usr/local/lib/ser/modules/xmlrpc.so>: /usr/local/lib/ser/modules/xmlrpc.so: cannot open shared object file: No such file or directory 0(5613) parse error (75,13-14): failed to load module 0(5613) set_mod_param_regex: No module matching <xmlrpc> found 0(5613) parse error (142,20-21): Can't set module parameter 0(5613) parse error (235,16-17): unknown command, missing loadmodule? ERROR: bad config file (4 errors) [root@ser ser]# ls -l /usr/local/lib/ser/modules/ total 4944 -rwxr-xr-x 1 root root 67012 Dec 17 15:02 acc_db.so -rwxr-xr-x 1 root root 73256 Dec 17 15:02 acc_syslog.so -rwxr-xr-x 1 root root 69954 Dec 17 15:02 auth_db.so -rwxr-xr-x 1 root root 58083 Dec 17 15:02 auth.so -rwxr-xr-x 1 root root 37353 Dec 17 15:02 avp_db.so -rwxr-xr-x 1 root root 168776 Dec 17 15:02 avpops.so -rwxr-xr-x 1 root root 79122 Dec 17 15:02 avp.so -rwxr-xr-x 1 root root 279393 Dec 17 15:02 ctl.so -rwxr-xr-x 1 root root 72908 Dec 17 15:02 db_ops.so -rwxr-xr-x 1 root root 63015 Dec 17 15:02 dispatcher.so -rwxr-xr-x 1 root root 21612 Dec 17 15:02 diversion.so -rwxr-xr-x 1 root root 114544 Dec 17 15:02 domain.so -rwxr-xr-x 1 root root 45712 Dec 17 15:02 enum.so -rwxr-xr-x 1 root root 84369 Dec 17 15:02 exec.so -rwxr-xr-x 1 root root 81406 Dec 17 15:02 fifo.so -rwxr-xr-x 1 root root 53450 Dec 17 15:02 flatstore.so -rwxr-xr-x 1 root root 43426 Dec 17 15:02 gflags.so -rwxr-xr-x 1 root root 87582 Dec 17 15:02 lcr.so -rwxr-xr-x 1 root root 35891 Dec 17 15:02 maxfwd.so -rwxr-xr-x 1 root root 112643 Dec 17 15:02 mediaproxy.so -rwxr-xr-x 1 root root 87493 Dec 17 15:02 msilo.so -rwxr-xr-x 1 root root 187830 Dec 17 15:02 nathelper.so -rwxr-xr-x 1 root root 21211 Dec 17 15:02 options.so -rwxr-xr-x 1 root root 131901 Dec 17 15:02 pdt.so -rwxr-xr-x 1 root root 272121 Dec 17 15:02 permissions.so -rwxr-xr-x 1 root root 122166 Dec 17 15:02 pike.so -rwxr-xr-x 1 root root 26180 Dec 17 15:02 print.so -rwxr-xr-x 1 root root 47605 Dec 17 15:02 ratelimit.so -rwxr-xr-x 1 root root 152851 Dec 17 15:02 registrar.so -rwxr-xr-x 1 root root 114904 Dec 17 15:02 rr.so -rwxr-xr-x 1 root root 63288 Dec 17 15:02 sanity.so -rwxr-xr-x 1 root root 66950 Dec 17 15:02 sl.so -rwxr-xr-x 1 root root 34423 Dec 17 15:02 speeddial.so -rwxr-xr-x 1 root root 120815 Dec 17 15:02 textops.so -rwxr-xr-x 1 root root 43284 Dec 17 15:02 timer.so -rwxr-xr-x 1 root root 848970 Dec 17 15:02 tm.so -rwxr-xr-x 1 root root 117605 Dec 17 15:02 uac.so -rwxr-xr-x 1 root root 38801 Dec 17 15:02 unixsock.so -rwxr-xr-x 1 root root 38987 Dec 17 15:02 uri_db.so -rwxr-xr-x 1 root root 38611 Dec 17 15:02 uri.so -rwxr-xr-x 1 root root 351480 Dec 17 15:02 usrloc.so -rwxr-xr-x 1 root root 100362 Dec 17 15:02 xlog.so [root@ser ser]# ************************************************************************************* The message is intended for the named addressee only and may not be disclosed to or used by anyone else, nor may it be copied in any way. The contents of this message and its attachments are confidential and may also be subject to legal privilege. If you are not the named addressee and/or have received this message in error, please advise us by e-mailing security(a)colt.net and delete the message and any attachments without retaining any copies. Internet communications are not secure and COLT does not accept responsibility for this message, its contents nor responsibility for any viruses. No contracts can be created or varied on behalf of COLT Telecommunications, its subsidiaries or affiliates ("COLT") and any other party by email Communications unless expressly agreed in writing with such other party. Please note that incoming emails will be automatically scanned to eliminate potential viruses and unsolicited promotional emails. For more information refer to www.colt.net or contact us on +44(0)20 7390 3900. ________________________________ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

Re: [Serusers] SIP Express Bundle - let's get started
by Mike Trest - Personal 17 Dec '07

17 Dec '07

At 09:10 AM 12/17/2007, Greger V. Teigre wrote: >Sounds good, Mike. >As we use berlios.de, there's really no official CVS maintainer, so >my comment is as good as any. That is a stable CVS. I only asked for maintainer as standard courtesy to be able to check-in ongoing documentation for the project if such materials are to be stored there. >Why do you need a mirror? Only offered if team see a need for a separate site for safety or separate "read-only" distribution overhead. >I'm sure we can create a new cvs module in the official repository >(though we may want to evaluate whether to use berlios' subversion instead). My best developers friends in US and CA are in various stages of moving from old CVS to subversion. It started as personal preference based on ease-of-use. Functionality of SVN is moving onward and correcting some traditional 'pain' associated with CVS when used by multiple active users. Client side tools available and growing too. >If we use subversion, you can use svk to create a mirror. Yup. "It supports repository mirroring, disconnected operation, history-sensitive merging, and integrates with other version control systems, as well as popular visual merge tools. " A PERL script to automate smart mirroring within your own subversion site was made available in 2005. I do not know how widely it is used. But my guys liked it as a way to be even more selective. http://subversion.tigris.org/servlets/ReadMsg?listName=users&msgNo=32596&at… >g-) > >Mike Trest - Personal wrote: >>Greger, et. al. >> >>While I did volunteer for Documentation and Testing already, >>I am going to branch out and perform each step myself >>so we can "document as we design". >> >>To that end I am setting up two "target machines" in New York on which >>to install-test-document each step wearing the hat of the Enterprise-IT-Guy. >>I have a couple of techs that we can use in the role of "outside tester" who >>will strictly follow the documentation to give the procedures a >>good Q-A test. >>I can be ready in early Janurary. >> >>I would like to hear from the current CVS maintainer regards creation >>of third machine as a Subversion / CVS mirror. Can you contact me >>via EMAIL with some guidance? >> >>Regards, ..mike.. >> >>At 03:03 AM 12/16/2007, Greger V. Teigre wrote: >>>Using kickstart sounds good to me. I think we should keep >>>separate the different steps: >>>1. Create a base image of the OS >>>2. Install the iptel.org apps and dependencies >>>3. Do local configurations >>>4. Create deployable virtual machine >>> >>>If each of these steps are isolated, different people can be >>>responsible for the different steps. >>> >>>I have reorganized the project page into sections and incorporated >>>lots of input from Jeremy at >>><http://iptel.org/bundle/project>http://iptel.org/bundle/project >>>We now need to create a project plan, split into responsibilities, >>>and take some key decisions. >>>I think one key decision that we have already converged on is: for >>>first version use CentOS5 and create vmware appliance. >>>g-) >>> >>> >>>Jai Rangi wrote: >>>>Sounds good, >>>>Here is the script. Its been a while that we have used it. We >>>>might need to modify something. I can look into it later. Let me >>>>know if this is something you are looking for. >>>>************************ >>>>#!/bin/sh >>>># Need to add /bin/sh path >>>># This script will install the softwares required to build softswitch. >>>># Sysrequirements FC5, DELL 1950, >>>># Jai >>>># 12/11/06 >>>> >>>>HOSTNAME=`hostname` >>>>TEMPLATE=<http://192.168.2.201/> 192.168.1.1 >>>> >>>>if [ "`id -u`" != "0" -o `hostname` != "$HOSTNAME" ]; then >>>> printf "NEED TO BE ROOT AND TO RUN ON $HOSTNAME\n" >>>> exit 0 >>>>fi >>>> >>>>echo "You are ROOT on Make sure you know what are you doing, >>>> Press ENTER to continue" >>>> >>>>read >>>> >>>>echo "Server name = $HOSTNAME" >>>>echo "FTP Server = $TEMPLATE" >>>>echo "Press ENTER to start" >>>> >>>>read >>>> >>>>printf "PROCEEDING WITH INSTALLATION OF Softswitch on ${HOSTNAME} ...\n\n" >>>> >>>> >>>>printf "ARE YOU SURE YOU WANT TO CONTINUE (Y/N)?" >>>>read X >>>>if [ "x$X" != "xY" ]; then >>>> printf "aborting. Press Y or N \n" >>>> exit 0 >>>>fi >>>>printf "\n" >>>> >>>>mkdir //home/switch >>>>ln -s /home/switch /usr/local/ >>>> >>>> >>>>cd /usr/local/src >>>> >>>>echo "------INSTALLING freeradius -----------------------------" >>>> >>>>printf "Do you want to INSTALL FREERADIUS (Y/N)?" >>>>read X >>>>if [ "x$X" == "xY" ]; then >>>>cd /usr/local/src/ >>>>wget >>>><ftp://ftp.freeradius.org/pub/radius/old/freeradius-1.1.2.tar.gz>ftp://ftp.freeradius.org/pub/radius/old/freeradius-1.1.2.tar.gz >>>> >>>>tar zxvf freeradius-1.1.2.tar.gz >>>>cd freeradius-1.1.2 >>>>./configure >>>> >>>>make >>>>make install >>>>#cd .. >>>>echo "enable digest auth in radiusd.conf, configure >>>>clients.conf, create radius db and import tables." >>>> >>>>else >>>>printf "OK YOU Pressed N that means you don't want free radius. >>>>No Problem \n" >>>>fi >>>> >>>>echo "------ radius client lib -----------------------------" >>>>printf "Do you want to radius client lib (Y/N)?" >>>> >>>>read X >>>> >>>>if [ "x$X" == "xY" ]; then >>>>cd /usr/local/src/ >>>>wget >>>><http://download.berlios.de/radiusclient-ng/radiusclient-ng-0.5.2.tar.gz>http://download.berlios.de/radiusclient-ng/radiusclient-ng-0.5.2.tar.gz >>>> >>>> >>>>tar zxvf radiusclient-ng-0.5.2.tar.gz >>>>cd radiusclient-ng-0.5.2 >>>>./configure >>>>make >>>>make install >>>> >>>>#vi /etc/ld.so.conf >>>># /usr/local/lib >>>>cat "/usr/local/lib" > /etc/ld.so.conf.d/radius- i386.conf >>>> >>>>ldconfig >>>>cd .. >>>> >>>>else >>>>printf "OK YOU did not Pressed Y that means you don't want radius >>>>client lib. I don't care, I am just a script \n" >>>> >>>>fi >>>> >>>>echo "------ MEDIAPROXY -------------------" >>>>printf "Do you want to Mediaproxy (Y/N)?" >>>> >>>>read X >>>> >>>>if [ "x$X" == "xY" ]; then >>>>cd /usr/local/src/ >>>>wget >>>><http://mediaproxy.ag-projects.com/old/mediaproxy-1.4.2.tar.gz>http://mediaproxy.ag-projects.com/old/mediaproxy-1.4.2.tar.gz >>>>tar mediaproxy-1.4.2.tar.gz >>>>mv mediaproxy /usr/local/mediaproxy >>>> >>>>#cp mediaproxy.ini.sample mediaproxy.ini >>>> >>>>#vi mediaproxy.ini >>>># edit the options as per our envi. >>>>cd /usr/local/mediaproxy/ >>>>wget <ftp://$TEMPLATE/configure_ser>ftp://$TEMPLATE/configure_ser >>>>_server/configfiles/mediaproxy.ini >>>> >>>>cp /usr/local/mediaproxy/boot/mediaproxy.other /etc/init.d/mediaproxy >>>>/etc/init.d/mediaproxy start >>>>cd /usr/local/src/ >>>> >>>>else >>>>printf "OK YOU did not Pressed Y that means you dont want >>>>Mediaproxy. I dont care, But remember SIP users wont be able to talk \n" >>>>fi >>>> >>>>echo "------ ser -------------------" >>>>printf "Do you want to INSTALL SER (Y/N)?" >>>> >>>>read X >>>> >>>>if [ "x$X" == "xY" ]; then >>>> >>>>## We make changes in the Makefile for all the modules, build our >>>>own tar and store it on our ftp server. >>>>#wget >>>><http://ftp.iptel.org/pub/ser/0.9.6/src/ser-0.9.6_src.tar.gz>http://ftp.iptel.org/pub/ser/0.9.6/src/ser-0.9.6_src.tar.gz >>>> >>>>cd /usr/local/src/ >>>>wget >>>><ftp://$TEMPLATE/configure_ser_server/ser-0.9.6_bingo.tar.gz>ftp://$TEMPLATE/configure_ser_server/ser-0.9.6_bingo.tar.gz >>>>echo "Press ENTER to start" >>>>read >>>>tar zxvf ser-0.9.6_bingo.tar.gz >>>>echo "Press ENTER to start" >>>>read >>>> >>>>cd ser-0.9.6 >>>>pwd >>>> >>>>#cd modules/acc >>>>#vi Makefile >>>>#remove comment from sql and radius lines >>>> >>>>#vi Makefile >>>>#remove *_radius from exclude list >>>> >>>>make all >>>>make install >>>>cd /usr/local/etc/ser/ >>>>wget >>>><ftp://$TEMPLATE/configure_ser_server/configfiles/ser.cfg>ftp://$TEMPLATE/configure_ser_server/configfiles/ser.cfg >>>> >>>>else >>>>printf "OK YOU did not Pressed Y that means you don't want a >>>>softswitch you. Just playing with me HUHH, OK \n" >>>>fi >>>> >>>># >>>><http://www.openser.org/docs/openser-radius-1.0.x.html>http://www.openser.org/docs/openser-radius-1.0.x.html >>>># ---------------------------------------------- >>>>echo "------ APACHE -------------------" >>>>printf "Do you want to INSTALL APACHE (Y/N)?" >>>>read X >>>>if [ "x$X" == "xY" ]; then >>>>cd /usr/local/src/ >>>>wget >>>><http://www.axint.net/apache/httpd/httpd-2.2.3.tar.gz>http://www.axint.net/apache/httpd/httpd-2.2.3.tar.gz >>>>tar zxvf httpd-2.2.3.tar.gz >>>>cd httpd-2.2.3 >>>>./configure --enable-shared=max --enable-module=rewrite >>>>--enable-module=so --with-apxs2 >>>>make >>>>make install >>>> >>>>else >>>>printf "OK YOU did not Pressed Y that means you dont want APACHE. >>>>You might need this for serweb. \n" >>>>fi >>>> >>>>echo "------ IMAP -------------------" >>>>printf "Do you want to INSTALL IMAP, (Y/N)?" >>>>read X >>>> >>>>if [ "x$X" == "xY" ]; then >>>> >>>>cd /usr/local/src/ >>>>cd /usr/local/src/ >>>>#wget >>>><http://www.mirrorservice.org/sites/tp.cac.washington.edu/imap/old/imap-2004…>http://www.mirrorservice.org/sites/tp.cac.washington.edu/imap/old/imap-2004… >>>> >>>>wget >>>><ftp://192.168.2.201/configure_ser_server/imap-2004c1.tar.Z>ftp://192.168.2.201/configure_ser_server/imap-2004c1.tar.Z >>>> >>>>tar zxf imap-2004c1.tar.Z >>>> >>>>cd imap-2004c1 >>>>make slx SSLTYPE=none >>>>cd c-client/ >>>>cp c-client.a /usr/lib >>>>cp rfc822.h mail.h linkage.h /usr/include/ >>>> >>>>else >>>>printf "OK YOU did not Pressed Y that means you dont want APACHE. >>>>You might need this for serweb. \n" >>>>fi >>>> >>>> >>>>echo "------ PHP 4.xx -------------------" >>>>printf "Do you want to INSTALL PHP 4.xx, If you dont install this >>>>then DONT complain if your serweb doesnt work (Y/N)?" >>>> >>>>read X >>>> >>>>if [ "x$X" == "xY" ]; then >>>> >>>>cd /usr/local/src/ >>>>wget >>>><http://museum.php.net/php4/php-4.3.10.tar.gz>http://museum.php.net/php4/php-4.3.10.tar.gz >>>>tar zxvf php-4.3.10.tar.gz >>>>cd php-4.3.10 >>>>./configure --with-apxs2=/usr/local/apache2/bin/apxs --with-mysql >>>>--with-imap >>>>make >>>>make install >>>> >>>>cp php.ini-dist /usr/local/lib/php.ini >>>>cd /usr/local/apache2/ >>>>wget >>>><ftp://$TEMPLATE/configure_ser_server/configfiles/httpd.conf>ftp://$TEMPLATE/configure_ser_server/configfiles/httpd.conf >>>> >>>> >>>>#vi /usr/local/apache2/conf/httpd.conf >>>># AddType application/x-httpd-php .php .phtml >>>># LoadModule php4_module libexec/libphp4.so >>>># >>>># modify the index.html line and add index.php to the list >>>># >>>># Listen 81 >>>> >>>>/usr/local/apache2/bin/apachectl start >>>> >>>>else >>>>printf "OK YOU did not Pressed Y that means you refused to >>>>install PHP. You will need this for serweb. \n" >>>>fi >>>> >>>>echo "------ PHP 4.xx -------------------" >>>>printf "Do you want to INSTALL PHP 4.xx, If you dont install this >>>>then DONT complain if your serweb doesnt work (Y/N)?" >>>>read X >>>> >>>>if [ "x$X" == "xY" ]; then >>>> >>>>cd /usr/local/src/ >>>> >>>>wget >>>><http://ftp.iptel.org/pub/serweb/serweb-0.9.6.tar.gz>http://ftp.iptel.org/pub/serweb/serweb-0.9.6.tar.gz >>>> >>>>tar zxvf serweb-0.9.6.tar.gz >>>> >>>>mv serweb-0.9.6 serweb >>>># next few line have been taken care of httpd.conf in apache installation. >>>> >>>>#vi /usr/local/apache2/conf/httpd.conf >>>> >>>>#Alias /serweb "/usr/local/serweb/html/" >>>> >>>>#<Directory "/usr/local/serweb/html"> >>>># AllowOverride None >>>># Options None >>>># Order allow,deny >>>># Allow from all >>>>#</Directory> >>>> >>>>cd /usr/local/serweb/config >>>> >>>>#vi config* >>>> modify all config files are per your env. >>>> >>>>#edit: >>>>#vi /usr/local/etc/ser/ser.cfg >>>>#fifo_mode=666 >>>> >>>>#Send daily missed calls >>>>#Crontab line: >>>>#5 0 * * * /usr/local/bin/php >>>>/usr/local/serweb/scripts/cron_job/send_daily_missed_calls.php >>>> >>>>#Server monitoring >>>>#Crontab line: >>>>#* * * * * /usr/local/bin/php >>>>/usr/local/serweb/scripts/cron_job/read_ser_moni.php >>>> >>>>else >>>>printf "OK YOU did not Pressed Y that means you refused to >>>>install PHP. You will need this for serweb. \n" >>>>fi >>>> >>>> >>>>echo "------ DIALIP_ADMIN -------------------" >>>>printf "Do you want to INSTALL DIALUP_ADMIN (Y/N)?" >>>>read X >>>> >>>>if [ "x$X" == "xY" ]; then >>>> >>>>cd /usr/local/src/ >>>>wget >>>><http://nchc.dl.sourceforge.net/sourceforge/dialup-admin/dialup_admin-1.62.t…>http://nchc.dl.sourceforge.net/sourceforge/dialup-admin/dialup_admin-1.62.t… >>>> >>>> >>>>tar zxvf dialup_admin- 1.62.tar.gz >>>>mv dialup_admin /usr/local/radiusadmin >>>> >>>>#vi /usr/local/apache2/conf/httpd.conf >>>> >>>>#Alias /radius "/usr/local/radiusadmin/htdocs/" >>>> >>>>#<Directory "/usr/local/radiusadmin/htdocs"> >>>># AllowOverride None >>>># Options None >>>># Order allow,deny >>>># Allow from all >>>>#</Directory> >>>> >>>>#cd /usr/local/radiusadmin/sql >>>># import all sql queries to radius database >>>>#cd /usr/local/radiusadmin/conf >>>># modify config files >>>> >>>> >>>>********************** >>>> >>>> >>>>On Dec 10, 2007 9:15 PM, ram >>>><<mailto:talk2ram@gmail.com>talk2ram(a)gmail.com > wrote: >>>> >>>>On Dec 10, 2007 11:02 PM, Jai Rangi >>>><<mailto:jprangi@gmail.com>jprangi(a)gmail.com> wrote: >>>>Preparing Kickstart file is a good idea. >>>>How about running a post installation script in the kickstart >>>>file. We have done this a lots on time in FC*. The process should >>>>be something like this, >>>>1. Build a default kickstart file that will install all the >>>>required packages for all the ser components. >>>>2. The kickstart file have an option to run a post installation script. >>>>3. Now we need to have one zip file of all the components we want >>>>to install related to ser. (ser, werbwer, sems, sipp, sppsak >>>>etc). That file can be placed either on the CDs or can be >>>>downloaded from somewhere. All the components can be packaged in >>>>the one zip file of they can be downloaded separately. >>>>4. The script can be interactive. >>>> >>>> >>>> >>>>yes that will be the idea, for the CD install >>>> >>>>after install of OS, we can run post install script to compile >>>>rest all required for >>>>SER ( or compile already for the same OS Distro and move them >>>>directly to the respective Directory) >>>> >>>> >>>> >>>> >>>>BTW I have a small postinstallation shell script like that, and I >>>>can share that. >>>> >>>> >>>>post to group, may be if require people may use. >>>> >>>>ram >>>> >>>> >>>>-Jai >>>> >>>> >>>>On Dec 10, 2007 3:42 AM, SIP <<mailto:sip@arcdiv.com> >>>>sip(a)arcdiv.com> wrote: >>>>Greger V. Teigre wrote: >>>> > [removed sems, semsdev, and serdev from cc] >>>> > >>>> > >>>> > SIP wrote: >>>> >> Something I noted about the project. While I am all for a >>>> virtual image >>>> >> for testing purposes and getting things up and running while we're >>>> >> creating this project, I think the ultimate goal should be something >>>> >> like an ISO that's ready to go on a server. There are many of >>>> us in the >>>> >> systems world who have not bought into the whole virtualisation >>>> >> marketing gimmick when it comes to system deployment, and for >>>> the casual >>>> >> home user who's less comfortable with using VMs for an actual >>>> production >>>> >> system (which is, to be honest, a lot of people), it might be >>>> easier in >>>> >> the long term to get everything the way we want it and then >>>> make an ISO >>>> >> that's ready to drop onto whatever machine with all the >>>> components we need. >>>> >> >>>> > Ok. When you say "ISO that's ready to drop onto", do you have a >>>> > particular procedure in mind? >>>> > >>>> > g-) >>>>Something like taking a kickstart file and making an installable livecd >>>>from it (something similar to the LiveCD berlios project). I don't KNOW >>>>of one that works to create installable ISOs in CentOS, but I'm sure >>>>there must be... somewhere... maybe....? >>>>N. >>>>_______________________________________________ >>>>Serusers mailing list >>>><mailto:Serusers@lists.iptel.org>Serusers(a)lists.iptel.org >>>>http://lists.iptel.org/mailman/listinfo/serusers >>>> >>>> >>>> >>>>_______________________________________________ >>>>Serusers mailing list >>>><mailto:Serusers@lists.iptel.org>Serusers(a)lists.iptel.org >>>>http://lists.iptel.org/mailman/listinfo/serusers >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>>_______________________________________________ >>>>Serusers mailing list >>>><mailto:Serusers@lists.iptel.org>Serusers(a)lists.iptel.org >>>> >>>>http://lists.iptel.org/mailman/listinfo/serusers >>>> >>> >>>_______________________________________________ >>>Serusers mailing list >>><mailto:Serusers@lists.iptel.org>Serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> >>_______________________________________________ >>Serusers mailing list >><mailto:Serusers@lists.iptel.org>Serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers >>

1 0

[Serusers] SER Video call GXV3000
by FaberK 17 Dec '07

17 Dec '07

Hi to All, I got a SER 0.8.14 up and running since more than 2 years. It's perfect. Now I'm testing 2 Grandstream GXV3000. They are able to register and make calls, but nothing to do with Video calls. I've tryied them on Asterisk, and they work. Video calls are great. I've notice that they use port 8002 for the video and on SER I got the rtpproxy listening on 8000, maybe is this the problem or is something else? Thanks to all. Have a nice day. -- .:FaberK:.

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[Serusers] SER 2.0 installation - SER does not start due to missing modules
by Strassburger, Patrick 17 Dec '07

17 Dec '07

Hello all, I am a newbie in SER and am failing since a few days installing SER 2.0. After several installation attempts I would like to ask you for some help. I did a "make group_include="standard mysql" all" but cannot see module mysql - where do I find those two modules and can I add them afterwards? Kind regards Patrick I get the following error and do not know how to resolve this: [root@ser ser]# ser -c 0(5613) ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: /usr/local/lib/ser/modules/mysql.so: cannot open shared object file: No such file or directory 0(5613) parse error (55,13-14): failed to load module 0(5613) ERROR: load_module: could not open module </usr/local/lib/ser/modules/xmlrpc.so>: /usr/local/lib/ser/modules/xmlrpc.so: cannot open shared object file: No such file or directory 0(5613) parse error (75,13-14): failed to load module 0(5613) set_mod_param_regex: No module matching <xmlrpc> found 0(5613) parse error (142,20-21): Can't set module parameter 0(5613) parse error (235,16-17): unknown command, missing loadmodule? ERROR: bad config file (4 errors) [root@ser ser]# ls -l /usr/local/lib/ser/modules/ total 4944 -rwxr-xr-x 1 root root 67012 Dec 17 15:02 acc_db.so -rwxr-xr-x 1 root root 73256 Dec 17 15:02 acc_syslog.so -rwxr-xr-x 1 root root 69954 Dec 17 15:02 auth_db.so -rwxr-xr-x 1 root root 58083 Dec 17 15:02 auth.so -rwxr-xr-x 1 root root 37353 Dec 17 15:02 avp_db.so -rwxr-xr-x 1 root root 168776 Dec 17 15:02 avpops.so -rwxr-xr-x 1 root root 79122 Dec 17 15:02 avp.so -rwxr-xr-x 1 root root 279393 Dec 17 15:02 ctl.so -rwxr-xr-x 1 root root 72908 Dec 17 15:02 db_ops.so -rwxr-xr-x 1 root root 63015 Dec 17 15:02 dispatcher.so -rwxr-xr-x 1 root root 21612 Dec 17 15:02 diversion.so -rwxr-xr-x 1 root root 114544 Dec 17 15:02 domain.so -rwxr-xr-x 1 root root 45712 Dec 17 15:02 enum.so -rwxr-xr-x 1 root root 84369 Dec 17 15:02 exec.so -rwxr-xr-x 1 root root 81406 Dec 17 15:02 fifo.so -rwxr-xr-x 1 root root 53450 Dec 17 15:02 flatstore.so -rwxr-xr-x 1 root root 43426 Dec 17 15:02 gflags.so -rwxr-xr-x 1 root root 87582 Dec 17 15:02 lcr.so -rwxr-xr-x 1 root root 35891 Dec 17 15:02 maxfwd.so -rwxr-xr-x 1 root root 112643 Dec 17 15:02 mediaproxy.so -rwxr-xr-x 1 root root 87493 Dec 17 15:02 msilo.so -rwxr-xr-x 1 root root 187830 Dec 17 15:02 nathelper.so -rwxr-xr-x 1 root root 21211 Dec 17 15:02 options.so -rwxr-xr-x 1 root root 131901 Dec 17 15:02 pdt.so -rwxr-xr-x 1 root root 272121 Dec 17 15:02 permissions.so -rwxr-xr-x 1 root root 122166 Dec 17 15:02 pike.so -rwxr-xr-x 1 root root 26180 Dec 17 15:02 print.so -rwxr-xr-x 1 root root 47605 Dec 17 15:02 ratelimit.so -rwxr-xr-x 1 root root 152851 Dec 17 15:02 registrar.so -rwxr-xr-x 1 root root 114904 Dec 17 15:02 rr.so -rwxr-xr-x 1 root root 63288 Dec 17 15:02 sanity.so -rwxr-xr-x 1 root root 66950 Dec 17 15:02 sl.so -rwxr-xr-x 1 root root 34423 Dec 17 15:02 speeddial.so -rwxr-xr-x 1 root root 120815 Dec 17 15:02 textops.so -rwxr-xr-x 1 root root 43284 Dec 17 15:02 timer.so -rwxr-xr-x 1 root root 848970 Dec 17 15:02 tm.so -rwxr-xr-x 1 root root 117605 Dec 17 15:02 uac.so -rwxr-xr-x 1 root root 38801 Dec 17 15:02 unixsock.so -rwxr-xr-x 1 root root 38987 Dec 17 15:02 uri_db.so -rwxr-xr-x 1 root root 38611 Dec 17 15:02 uri.so -rwxr-xr-x 1 root root 351480 Dec 17 15:02 usrloc.so -rwxr-xr-x 1 root root 100362 Dec 17 15:02 xlog.so [root@ser ser]# ************************************************************************************* The message is intended for the named addressee only and may not be disclosed to or used by anyone else, nor may it be copied in any way. The contents of this message and its attachments are confidential and may also be subject to legal privilege. If you are not the named addressee and/or have received this message in error, please advise us by e-mailing security(a)colt.net and delete the message and any attachments without retaining any copies. Internet communications are not secure and COLT does not accept responsibility for this message, its contents nor responsibility for any viruses. No contracts can be created or varied on behalf of COLT Telecommunications, its subsidiaries or affiliates ("COLT") and any other party by email Communications unless expressly agreed in writing with such other party. Please note that incoming emails will be automatically scanned to eliminate potential viruses and unsolicited promotional emails. For more information refer to www.colt.net or contact us on +44(0)20 7390 3900.

2 1

[OpenSER-Users] IMC Module
by Amit Vijayvargiya 17 Dec '07

17 Dec '07

Hi All, I am using openser as a proxy and presence server. now i am add new module IMC for support chat conference. But how i can test changes are right???????? There is any standard phone Or sip client which support IMC module and able to send messages like create,join.... for conference. So, i can test our sip proxy........... Thanks & Regards, --Amit

2 6

[OpenSER-Users] OpenSER Oracle
by Martin.Gruber1＠infineon.com 17 Dec '07

17 Dec '07

HI, How did you solve the problem....Suddenly I also got an error: [S1000][unixODBC][Oracle][ODBC][Ora]ORA-12514: TNS:listener does not currently know of service requested in connect descriptor Does anybody know the problem? Regards Martin

2 1

[OpenSER-Users] xmpp error
by TCB 17 Dec '07

17 Dec '07

i'm trying to setup openser-1.3 with xmpp support . I get the following error when I to IM another user. :cmd_send_message: invalid content-type 0xff00f -- TC

2 1

[OpenSER-Users] Help regarding tls
by Vijender Singh, Noida 17 Dec '07

17 Dec '07

Hi Sir, For enabling tls support on openser version 1.0.1 on Fedora 7, I have the attached openser.cfg file which I am using. tls_certificate = "/sip-server/tls/tools/user/user/user-cert.pem" tls_private_key = "/sip-server/tls/tools/user/user-privkey.pem" tls_ca_list = "/sip-server/tls/tools/user/user-calist.pem" The three given certificates are generated using scripts gen_rootCA.sh and gen_usercert.sh. When I try to run openser, it gives me the error saying "ERROR: bad config file (4 errors)" I have done configuration on the basis of the instruction given in the link http://www.openser.org/docs/tls.html#AEN93. I also installed the openssl-0.9.8g on my Fedora machine. Can you help me out as I need to complete my work on very high priority? Thanks in advance Vijender DISCLAIMER: ----------------------------------------------------------------------------------------------------------------------- The contents of this e-mail and any attachment(s) are confidential and intended for the named recipient(s) only. It shall not attach any liability on the originator or HCL or its affiliates. Any views or opinions presented in this email are solely those of the author and may not necessarily reflect the opinions of HCL or its affiliates. Any form of reproduction, dissemination, copying, disclosure, modification, distribution and / or publication of this message without the prior written consent of the author of this e-mail is strictly prohibited. If you have received this email in error please delete it and notify the sender immediately. Before opening any mail and attachments please check them for viruses and defect. -----------------------------------------------------------------------------------------------------------------------

2 1

[OpenSER-Users] Stored Procedure
by Il Neofita 17 Dec '07

17 Dec '07

Hi, I am looking to openser 1.3 and I cannot figure out how to use the mysql stored procedure. Anyone, can post a little example?

1 0

[OpenSER-Users] multi-domain support
by satish patel 17 Dec '07

17 Dec '07

This is my config file debug=3 #fork=no log_stderror=yes listen=192.168.104.20 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" 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/domain.so" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" modparam("domain", "db_url", "mysql://root:tulip@localhost/ser") #modparam("domain", "domain_table", "domain") #modparam("domain", "domain_col", "domain") modparam("usrloc|registrar|auth_db|avpops|group", "use_domain", 1) modparam("domain", "db_mode", 1) # --- main routing logic route{ 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; }; if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); }; if (is_uri_host_local() || uri==myself) { if (method=="REGISTER") { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; }; if (is_uri_host_local() || uri==myself) { if (method=="REGISTER") { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); break; }; if (is_from_local() || from_uri==myself) { if (method=="REGISTER") { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; }; if (is_uri_host_local() || uri==myself) { if (method=="REGISTER") { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); break; }; save("location"); exit; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; exit; } when i run ser i got error modules error . is it correct or not can u check it ??? ----PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org --------------------------------- Never miss a thing. Make Yahoo your homepage.

2 1

[OpenSER-Users] [LCR] how does "to_gw()" work?
by Iñaki Baz Castillo 17 Dec '07

17 Dec '07

Hi, I read in LCR doc: 1.4.4. to_gw([group-id]) Checks if in-dialog request goes to a gateway. If an optional group-id is given, only gateways belonging to this group are checked. I suppose that a gateway uses its scheme:IP:port as "Contact" in 200 OK, so "to_gw()" function matches that scheme:IP:port in RURI and verifies if there is a To tag to match an in-dialog message and deliver it to gateway. Is it? Thanks for any explanation. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

2 3

[OpenSER-Users] [Fwd: Re: IMC Module]
by Amit Vijayvargiya 17 Dec '07

17 Dec '07

1 0

[OpenSER-Users] Any linux/windows/mac os SIPs client to use with OpenSER-tls ?
by Marc LEURENT 17 Dec '07

17 Dec '07

Good afternoon, Have you got names of SIP client with sip-tls support that you have seen working with OpenSER? Thanks I'm asking that before having to deploy a WPA2-Enterprise wireless network to ensure masked signaling / (and perhaps masked communications)

2 1

[Serusers] multi-domain support
by satish patel 17 Dec '07

17 Dec '07

Dear all i have going to install ser with multidomain support like aaa(a)example.com bbb(a)openser.org ccc(a)iptel.org Something like this but i done know how to configure it in CFG file is there anybody have runing configuration or sample example file plz hint me and send it me .... $ cat ~/satish/url.txt http://www.linuxbug.org _____________________________________________________________________________________________________ --------------------------------- Chat on a cool, new interface. No download required. Click here.

2 1

[OpenSER-Users] Allow_trusted problems
by CSB 17 Dec '07

17 Dec '07

For some reason I cannot get allow_trusted() to evaluate to true for a matching URI. Relevant sections of the config (OpenSER 1.1): modparam("auth_db|permissions|uri_db|usrloc|domain", "db_url", "mysql://username:password@localhost/ser") modparam("permissions", "db_mode", 0) modparam("permissions", "trusted_table", "trusted") route { if (!allow_trusted()) { xlog("!allow_trusted. rs $rs si $si rm $rm ru $ru tu $tu fu $fu rr $rr"); }; } The trusted table contains the following: +----------------+-------+--------------+------+ | src_ip | proto | from_pattern | tag | +----------------+-------+--------------+------+ | 202.200.20.200 | any | NULL | NULL | +----------------+-------+--------------+------+ 1 row in set (0.00 sec) I can see that the following query is being executed: select proto,from_pattern,tag from trusted where src_ip='202.200.20.200' But no match is made because the log file shows: Dec 17 15:37:56 beta /usr/local/sbin/openser[7209]: !allow_trusted. rs <null> si 147.200.20.200 rm INVITE ru sip:5551234@147.200.20.200 tu sip:5551234@147.200.20.200 fu sip:021446202@202.200.20.200 rr <null> Could anyone provide any suggestions on what I'm doing wrong? Regards Cameron

2 5

[OpenSER-Users] [Fwd: IMC Module]
by Amit Vijayvargiya 17 Dec '07

17 Dec '07

1 0

[OpenSER-Users] OpenSER billing options
by Andy Smith 17 Dec '07

17 Dec '07

Hi list, are there alternatives to Radius for billing integration with OpenSER? The problem Radius seems to pose me is that it is per user and therefore requires to have a full database of every softphone/SNOM/etc user, which is admin intensive. Does anyone know if there is a method of accounting data per SIP peer (for example per Asterisk system attached to OpenSER) which could record call details without needing to know about all the actual handset users? I suppose you can do anything you want with the accounting module so this is really a "does anyone already do this" question so I dont need to invent everything from scratch... thanks for any suggestions, Andy.

2 1

[OpenSER-Users] Problem Registering via TCP
by Krom 16 Dec '07

16 Dec '07

I have a UA that I am attempting to register to OpenSER 1.1.1 via TCP and the only hint I can give it the following statement that is placed in the syslog file. WARNING: io_wait_loop_sigio_rt: ignoring event 41 on fd 0 (fm->fd=0, fm->data=(nil)) I have placed in the first line of the main route the following just to see if the registration is making it to the main route and it is not. if(proto==TCP) { xlog("L_INFO","SIP message was received over TCP\n"); }; I can tell from tcpdump that the packet is being received and a complete SYN,ACK is successful. Global Config: listen=10.1.1.28:5060 disable_tls=1 tcp_children=4 tcp_send_timeout=3 tcp_connect_timeout=5 tcp_connection_lifetime=3600 A netsat shows the port in a listening state. When starting OpenSER it shows that it is listening on TCP:IP_ADDR:5060 When the UA sends a register request, a connection is established, the one statement above is placed in the log file and then the connection goes to a CLOSE_WAIT state. Any suggestions. I know at least one of us has seen this before. I searched the list archives and found one thread in Nov 06 discussing the same issue; but no follow-up of resolution posted. http://lists.openser.org/pipermail/users/2006-November/007653.html -- -- Krom kyle.romulas(a)gmail.com

1 1

[OpenSER-Users] CPL doesn't allow "redirect" to url="http://webpage"
by Iñaki Baz Castillo 16 Dec '07

16 Dec '07

Hi, I want to set a redirect to a web page in CPL but I get error: <cpl> <incoming> <lookup source="registration"> <success> <proxy /> </success> <notfound> <location url="http://google.es"> <redirect /> </location> </notfound> </lookup> </incoming> </cpl> ~# openserctl fifo LOAD_CPL sip:user@domain /etc/openser/cpl/user(a)domain.xml 500 Bad CPL file Log:: Error: Encoding of the CPL script failed It works if I set (as usual): <location url="sip:user2@domain"> Why doesn't CPL module allow redirection to non SIP target? -- Iñaki Baz Castillo

1 0

[OpenSER-Users] OpenSER with multidomain support
by satish patel 15 Dec '07

15 Dec '07

Dear all I have post many question about multidomain support help for openser i need cfg file which you ppl are useing caz openser.cfg is bit harder for me and i have also find many help on ser on google but still i m confuse about parmeter and options if anybody have done this configuration send it me or explain me how to do it ----PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org --------------------------------- Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.

2 1

[OpenSER-Users] get Count of items in lists
by Dave Singer 14 Dec '07

14 Dec '07

using openser 1.3, is there a way to get a count of how many items there are in an avp variable other than: while ( is_avp_set("$avp(s:fork_uri)") ) { $var(fork_uri_count) = $var(fork_uri_count) + 1 ; avp_copy("$avp(s:fork_uri)", "$avp(s:fork_uri_rev_order)/d") ; } then if you wanted things in the same order as they were originally you would also have to: while ( is_avp_set("$avp(s:fork_uri_rev_order)") ) { avp_copy("$avp(s:fork_uri_rev_order)", "$avp(s:fork_uri)/d") ; } talk about an expensive counting Several ways I could see this implimented: New function avp_count("$avp(s:myvar)", "$avp(s:count_result)") or avp_count("$avp(s:myvar)") where $rc contains the result. is_avp_set("$avp(s:myvar)") where $rc is set to the count, probably best as you could test if exitst and the go right to processing. avp_check("$avp(s:myvar)", "cn/g") where $rc is set to the count. It could always set $rc to count of found matches for when the g flag is used. Of course using $rc for is_avp_set() or avp_check() depends on if setting that to non-zero makes a test like if( is_avp_set("$avp(s:myvar)") ) always trigger as false. also on getting a count ... another place I need to get a count is how many times a character(s) (or shoot the moon, a regex) occurs in a string. for handeling things like transformations like {s.select,index,seperator}, eg: $(var(findme){ s.select,$var(count),,}) sidenote ... (not sure what values for seperator are valid) a transformation like {s.count,seperator} Here is an example of something I was trying to do that would have benifited from knowing the count of seperator. I think that I got a warning or error when the test for the while indexed past the end of the string. xlog("L_INFO", "Original findme list to parse: $avp(s:findme)\n"); if ( $avp(s:findme) =~ "\|" ) { $var(findme) = $(avp(s:findme){s.select,0,|}) ; $var(findme_tm_out) = $(avp(s:findme){s.select,1,|}) ; xlog("L_INFO", "This findme has a timeout: $var(findme_tm_out)\n"); } else { $var(findme) = $avp(s:findme) ; } xlog("L_INFO", "findme list to parse: $var(findme)\n"); $var(findme_cnt) = 0 ; while ($(var(findme){s.select,$var(findme_cnt),,}{s.len }) > 0 ) { xlog("L_INFO", "Loop $var(findme_cnt), Adding $(var(findme){s.select,$var(findme_cnt),,}) to TO_list\n"); $avp(s:TO_list) = $(var(findme){s.select,$var(findme_cnt),,}) ; $var(findme_cnt) = $var(findme_cnt) + 1 ; } can/do $var()'s stack like $avp()'s? Thanks Dave

2 4

Re: [OpenSER-Users] Missed calls (I?aki Baz Castillo)
by David Villasmil Govea 14 Dec '07

14 Dec '07

te refieres a ésto? ya está en el openser.cfg if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; > > > El Friday 14 December 2007 13:55:01 David Villasmil Govea escribi?: > > * Furthermore,?Openser doesn't see all INVITES+BYE, so i can't make > cdrs!! > > I've read somewhere that to log all transaction messages i should have > > "strict route" as oppossed to "loose route". > > No, you need to add "Record-Route" header. This is done by > "record_route()" > function of rr module. > > -- DVG -- Imagination is more important than knowledge Albert Einstein

1 0

Re: [OpenSER-Users] Users Digest, Vol 31, Issue 46
by David Villasmil Govea 14 Dec '07

14 Dec '07

te refieres a ésto? ya está en el openser.cfg if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; > > > El Friday 14 December 2007 13:55:01 David Villasmil Govea escribi?: > > * Furthermore,?Openser doesn't see all INVITES+BYE, so i can't make > cdrs!! > > I've read somewhere that to log all transaction messages i should have > > "strict route" as oppossed to "loose route". > > No, you need to add "Record-Route" header. This is done by > "record_route()" > function of rr module. > > -- DVG -- Imagination is more important than knowledge Albert Einstein

1 0

[OpenSER-Users] openser config issue or asterisk bug
by Patrick Baker 14 Dec '07

14 Dec '07

Internal UA's connecting to the sip proxy work fine, but as soon an External UA connects to the sip proxy via a NAT'd public IP address - the call lasts for approx 20 seconds then disconnects. From what I can see openser is attempting to respond to asterisk after the session is initiated via the public IP NAT'd address - and is unable since asterisk and openser are behind the firewall on the internal network ( ELBOW routing ). Any help getting this working would be most appreciated, I would also like to document the issue to help others that encounter similar problems as mine. External phones: UA -> [(public IP) Firewall (NAT to localip)] -> openser -> asterisk Internal phones: UA -> openser -> asterisk #### Layout ( asterisk and openser are on the same machine ) NATD public ip = xxx.206.xxx.136 -> 10.3.1.33 phone ip = xxx.206.xxx.137 openser = 10.3.1.33 p5060 asteriskSIP = 10.3.1.34 p5060 #### Asterisk SIP Config [general] canreinvite=no externip=xxx.206.xxx.136 localnet=10.3.1.0/255.255.255.0 context=default bindport=5060 bindaddr=10.3.1.34 sipdebug=yes [openser] type=friend context=default insecure=very externalnotify=yes allow=all host=10.3.1.33 #### SER LOG Dec 13 22:15:03 phonesys-slave openser[15976]: New request - M=INVITE RURI=sip:500@xxx.206.xxx.136:5060;user=phone F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=xxx.206.xxx.137 ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 Dec 13 22:15:03 phonesys-slave openser[15976]: Callee is not local - M=INVITE RURI=sip:500@xxx.206.xxx.136:5060;user=phone F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=xxx.206.xxx.137 ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 Dec 13 22:15:03 phonesys-slave openser[15976]: Request leaving server, D-URI='<null>' - M=INVITE RURI=sip:500@10.3.1.34:5060;transport=udp F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=xxx.206.xxx.137 ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 Dec 13 22:15:03 phonesys-slave openser[15973]: Reply - S=100 D=Trying F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=10.3.1.34 ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 Dec 13 22:15:03 phonesys-slave openser[15978]: Reply - S=200 D=OK F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=10.3.1.34 ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 Dec 13 22:15:04 phonesys-slave openser[15989]: Reply - S=200 D=OK F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=10.3.1.34 ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 Dec 13 22:15:05 phonesys-slave openser[15986]: Reply - S=200 D=OK F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=10.3.1.34 ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 Dec 13 22:15:07 phonesys-slave openser[15976]: Reply - S=200 D=OK F=sip:pbaker@xxx.206.xxx.136 T=sip:500@xxx.206.xxx.136;user=phone IP=10.3.1.34 ID=f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 #### Asterisk LOG *CLI> [Dec 13 22:15:03] <--- SIP read from 10.3.1.33:5060 ---> INVITE sip:500@10.3.1.34:5060;transport=udp SIP/2.0 Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804> Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804 To: <sip:500@xxx.206.xxx.136;user=phone> CSeq: 2 INVITE Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 Contact: <sip:pbaker@xxx.206.xxx.137> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.0.2708 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 69 Content-Type: application/sdp Content-Length: 254 v=0 o=- 978310717 978310717 IN IP4 xxx.206.xxx.137 s=Polycom IP Phone c=IN IP4 xxx.206.xxx.137 t=0 0 m=audio 35022 RTP/AVP 0 8 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> [Dec 13 22:15:03] --- (16 headers 11 lines) --- [Dec 13 22:15:03] == Using TOS bits 0 [Dec 13 22:15:03] == Using CoS mark 5 [Dec 13 22:15:03] Sending to 10.3.1.33 : 5060 (no NAT) [Dec 13 22:15:03] Using INVITE request as basis request - f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 [Dec 13 22:15:03] No user 'pbaker' in SIP users list [Dec 13 22:15:03] Found peer 'openser' for 'pbaker' from 10.3.1.33:5060 [Dec 13 22:15:03] Found RTP audio format 0 [Dec 13 22:15:03] Found RTP audio format 8 [Dec 13 22:15:03] Found RTP audio format 18 [Dec 13 22:15:03] Found RTP audio format 101 [Dec 13 22:15:03] Peer audio RTP is at port xxx.206.xxx.137:35022 [Dec 13 22:15:03] Found audio description format PCMU for ID 0 [Dec 13 22:15:03] Found audio description format PCMA for ID 8 [Dec 13 22:15:03] Found audio description format G729 for ID 18 [Dec 13 22:15:03] Found audio description format telephone-event for ID 101 [Dec 13 22:15:03] Capabilities: us - 0x27f9fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|t140), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) [Dec 13 22:15:03] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Dec 13 22:15:03] Peer audio RTP is at port xxx.206.xxx.137:35022 [Dec 13 22:15:03] Looking for 500 in default (domain 10.3.1.34) [Dec 13 22:15:03] list_route: hop: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804> [Dec 13 22:15:03] <--- Transmitting (no NAT) to 10.3.1.33:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804> From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804 To: <sip:500@xxx.206.xxx.136;user=phone> Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@10.3.1.34> Content-Length: 0 <------------> [Dec 13 22:15:03] -- Executing [500@default:1] Playback("SIP/xxx.206.xxx.136-090b5380", "demo-abouttotry") in new stack [Dec 13 22:15:03] Audio is at 10.3.1.34 port 47068 [Dec 13 22:15:03] Adding codec 0x4 (ulaw) to SDP [Dec 13 22:15:03] Adding codec 0x8 (alaw) to SDP [Dec 13 22:15:03] Adding codec 0x100 (g729) to SDP [Dec 13 22:15:03] Adding non-codec 0x1 (telephone-event) to SDP [Dec 13 22:15:03] <--- Reliably Transmitting (no NAT) to 10.3.1.33:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804> From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804 To: <sip:500@xxx.206.xxx.136;user=phone>;tag=as26a99907 Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@10.3.1.34> Content-Type: application/sdp Content-Length: 333 v=0 o=root 45593872 45593872 IN IP4 10.3.1.34 s=Asterisk PBX SVN-trunk-r92779 c=IN IP4 10.3.1.34 t=0 0 m=audio 47068 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Dec 13 22:15:04] -- <SIP/xxx.206.xxx.136-090b5380> Playing 'demo-abouttotry.slin' (language 'en') [Dec 13 22:15:04] Retransmitting #1 (no NAT) to 10.3.1.33:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804> From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804 To: <sip:500@xxx.206.xxx.136;user=phone>;tag=as26a99907 Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@10.3.1.34> Content-Type: application/sdp Content-Length: 333 v=0 o=root 45593872 45593872 IN IP4 10.3.1.34 s=Asterisk PBX SVN-trunk-r92779 c=IN IP4 10.3.1.34 t=0 0 m=audio 47068 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 13 22:15:05] Retransmitting #2 (no NAT) to 10.3.1.33:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804> From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804 To: <sip:500@xxx.206.xxx.136;user=phone>;tag=as26a99907 Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@10.3.1.34> Content-Type: application/sdp Content-Length: 333 v=0 o=root 45593872 45593872 IN IP4 10.3.1.34 s=Asterisk PBX SVN-trunk-r92779 c=IN IP4 10.3.1.34 t=0 0 m=audio 47068 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 13 22:15:07] Retransmitting #3 (no NAT) to 10.3.1.33:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804> From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804 To: <sip:500@xxx.206.xxx.136;user=phone>;tag=as26a99907 Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@10.3.1.34> Content-Type: application/sdp Content-Length: 333 v=0 o=root 45593872 45593872 IN IP4 10.3.1.34 s=Asterisk PBX SVN-trunk-r92779 c=IN IP4 10.3.1.34 t=0 0 m=audio 47068 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 13 22:15:11] Retransmitting #4 (no NAT) to 10.3.1.33:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804> From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804 To: <sip:500@xxx.206.xxx.136;user=phone>;tag=as26a99907 Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@10.3.1.34> Content-Type: application/sdp Content-Length: 333 v=0 o=root 45593872 45593872 IN IP4 10.3.1.34 s=Asterisk PBX SVN-trunk-r92779 c=IN IP4 10.3.1.34 t=0 0 m=audio 47068 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 13 22:15:15] Retransmitting #5 (no NAT) to 10.3.1.33:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804> From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804 To: <sip:500@xxx.206.xxx.136;user=phone>;tag=as26a99907 Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@10.3.1.34> Content-Type: application/sdp Content-Length: 333 v=0 o=root 45593872 45593872 IN IP4 10.3.1.34 s=Asterisk PBX SVN-trunk-r92779 c=IN IP4 10.3.1.34 t=0 0 m=audio 47068 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 13 22:15:18] -- Executing [500@default:2] Dial("SIP/xxx.206.xxx.136-090b5380", "IAX2/guest@misery.digium.com/s@default") in new stack [Dec 13 22:15:18] -- Called guest@misery.digium.com/s@default [Dec 13 22:15:18] -- Call accepted by 216.207.245.8 (format gsm) [Dec 13 22:15:18] -- Format for call is gsm [Dec 13 22:15:18] -- IAX2/216.207.245.8:4569-1 is ringing [Dec 13 22:15:19] -- IAX2/216.207.245.8:4569-1 answered SIP/xxx.206.xxx.136-090b5380 [Dec 13 22:15:19] Retransmitting #6 (no NAT) to 10.3.1.33:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.1.33;branch=z9hG4bK2fc1.94c7bb96.0;received=10.3.1.33 Via: SIP/2.0/UDP xxx.206.xxx.137;rport=5060;branch=z9hG4bKcbd9a65f8D4E5FE0 Record-Route: <sip:10.3.1.33;lr;ftag=CFE2AAA1-5B882804> From: "pbaker" <sip:pbaker@xxx.206.xxx.136>;tag=CFE2AAA1-5B882804 To: <sip:500@xxx.206.xxx.136;user=phone>;tag=as26a99907 Call-ID: f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 CSeq: 2 INVITE User-Agent: Asterisk PBX SVN-trunk-r92779 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:500@10.3.1.34> Content-Type: application/sdp Content-Length: 333 v=0 o=root 45593872 45593872 IN IP4 10.3.1.34 s=Asterisk PBX SVN-trunk-r92779 c=IN IP4 10.3.1.34 t=0 0 m=audio 47068 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Dec 13 22:15:23] WARNING[16421]: chan_sip.c:2337 retrans_pkt: Maximum retries exceeded on transmission f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 for seqno 2 (Critical Response) [Dec 13 22:15:23] WARNING[16421]: chan_sip.c:2364 retrans_pkt: Hanging up call f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137 - no reply to our critical packet. [Dec 13 22:15:23] -- Hungup 'IAX2/216.207.245.8:4569-1' [Dec 13 22:15:23] == Spawn extension (default, 500, 2) exited non-zero on 'SIP/xxx.206.xxx.136-090b5380' [Dec 13 22:15:24] Really destroying SIP dialog 'f7232b5-5d4b9a93-e6ff617e(a)xxx.206.xxx.137' Method: INVITE #### PCAP Diagram |Time | xxx.206.xxx.137 | 10.3.1.33 | |5.056 | Request: INVITE sip |SIP/SDP: Request: INVITE sip:500@xxx.206.xxx.136:5060;user=phone, with session description | |(5060) ------------------> (5060) | |5.058 | Status: 100 Trying |SIP: Status: 100 Trying | |(5060) <------------------ (5060) | |5.058 | Status: 407 Proxy A |SIP: Status: 407 Proxy Authentication Required | |(5060) <------------------ (5060) | |5.140 | Request: ACK sip:50 |SIP: Request: ACK sip:500@xxx.206.xxx.136:5060 | |(5060) ------------------> (5060) | |5.165 | Request: INVITE sip |SIP/SDP: Request: INVITE sip:500@xxx.206.xxx.136:5060;user=phone, with session description | |(5060) ------------------> (5060) | |5.165 | Status: 100 Trying |SIP: Status: 100 Trying | |(5060) <------------------ (5060) | |5.168 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session description | |(5060) <------------------ (5060) | |6.169 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session description | |(5060) <------------------ (5060) | |7.168 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session description | |(5060) <------------------ (5060) | |9.168 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session description | |(5060) <------------------ (5060) | |13.169 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session description | |(5060) <------------------ (5060) | |17.169 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session description | |(5060) <------------------ (5060) | |21.169 | Status: 200 OK, wit |SIP/SDP: Status: 200 OK, with session description | |(5060) <------------------ (5060) | #### OpenSER Config 1.2 ######################################################################## # This configuration is autogenerated by sip:wizard # (http://www.sipwise.com/wizard) on Thu Dec 13 22:43:43 +0100 2007 # for OpenSER 1.2 # # Copyright (C) 2007 Sipwise (support(a)sipwise.com) ######################################################################## ######################################################################## # By obtaining, using, and/or copying this configuration and/or its # associated documentation, you agree that you have read, understood, # and will comply with the Terms of Usage provided at # http://www.sipwise.com/news/?page_id=6 as well as the following # additions: # # Permission to use, copy, modify, and distribute this configuration and # its associated documentation for any purpose and without fee is hereby # granted, provided that the above copyright notice appears in all # copies, and that both that copyright notice and this permission notice # appear in supporting documentation, and that the name of Sipwise or # the author will not be used in advertising or publicity pertaining to # distribution of the configuration without specific, written prior # permission. ######################################################################## ######################################################################## # Before using this configuration, read the following prerequisites in # order to gain the designated functionallity: # # base: # You have to insert all locally served domains (i.e. # "openserctl domain add your.domain.com"). # # nat-rtpproxy: # You have to install RTPProxy # (http://www.openser.org/downloads/snapshots/rtpproxy/) for relaying # RTP traffic. # # offnet-pstn: # You have to add a routing entry for lcr (i.e. "openserctl lcr # addroute '' '' 1 1"). Additionally, you have to add your gateways # (i.e. "openserctl lcr addgw my-test-gw 1.2.3.4 5060 sip udp 1"). # ######################################################################## ######################################################################## # Configuration 'sip:wizard - Thu Dec 13 22:43:43 +0100 2007' ######################################################################## listen = udp:10.3.1.33:5060 mpath = "/usr/local/lib/openser/modules" children = 8 debug = 3 fork = yes group = "openser" user = "openser" disable_tcp = no log_facility = LOG_DAEMON log_stderror = no tcp_children = 4 mhomed = no server_signature = yes sock_group = "openser" sock_mode = 0600 sock_user = "openser" unix_sock = "/tmp/openser.sock" unix_sock_children = 1 reply_to_via = no sip_warning = yes check_via = no dns = no rev_dns = no disable_core_dump = no dns_try_ipv6 = yes dns_use_search_list = yes loadmodule "usrloc.so" modparam("usrloc", "user_column", "username") modparam("usrloc", "domain_column", "domain") modparam("usrloc", "contact_column", "contact") modparam("usrloc", "expires_column", "expires") modparam("usrloc", "q_column", "q") modparam("usrloc", "callid_column", "callid") modparam("usrloc", "cseq_column", "cseq") modparam("usrloc", "methods_column", "methods") modparam("usrloc", "flags_column", "flags") modparam("usrloc", "user_agent_column", "user_agent") modparam("usrloc", "received_column", "received") modparam("usrloc", "socket_column", "socket") modparam("usrloc", "use_domain", 0) modparam("usrloc", "desc_time_order", 0) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("usrloc", "db_mode", 1) modparam("usrloc", "matching_mode", 0) modparam("usrloc", "cseq_delay", 20) modparam("usrloc", "nat_bflag", 6) loadmodule "textops.so" loadmodule "rr.so" modparam("rr", "enable_full_lr", 0) modparam("rr", "append_fromtag", 1) modparam("rr", "enable_double_rr", 1) modparam("rr", "add_username", 0) loadmodule "tm.so" modparam("tm", "fr_timer", 30) modparam("tm", "fr_inv_timer", 120) modparam("tm", "wt_timer", 5) modparam("tm", "delete_timer", 2) modparam("tm", "noisy_ctimer", 0) modparam("tm", "ruri_matching", 1) modparam("tm", "via1_matching", 1) modparam("tm", "unix_tx_timeout", 2) modparam("tm", "restart_fr_on_each_reply", 1) modparam("tm", "pass_provisional_replies", 0) loadmodule "xlog.so" modparam("xlog", "buf_size", 4096) modparam("xlog", "force_color", 0) loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("mi_fifo", "fifo_mode", 0660) modparam("mi_fifo", "fifo_group", "openser") modparam("mi_fifo", "fifo_user", "openser") modparam("mi_fifo", "reply_dir", "/tmp/") modparam("mi_fifo", "reply_indent", "\t") loadmodule "domain.so" modparam("domain", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("domain", "db_mode", 1) modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 60) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "rtpproxy_disable", 0) modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "received_avp", "$avp(i:801)") loadmodule "sl.so" modparam("sl", "enable_stats", 1) loadmodule "uri.so" loadmodule "registrar.so" modparam("registrar", "default_expires", 3600) modparam("registrar", "min_expires", 60) modparam("registrar", "max_expires", 0) modparam("registrar", "default_q", 0) modparam("registrar", "append_branches", 1) modparam("registrar", "case_sensitive", 0) modparam("registrar", "received_param", "received") modparam("registrar", "max_contacts", 0) modparam("registrar", "retry_after", 0) modparam("registrar", "method_filtering", 0) modparam("registrar", "path_mode", 2) modparam("registrar", "path_use_received", 0) modparam("registrar", "received_avp", "$avp(i:801)") loadmodule "maxfwd.so" modparam("maxfwd", "max_limit", 256) loadmodule "mysql.so" modparam("mysql", "ping_interval", 300) modparam("mysql", "auto_reconnect", 1) loadmodule "auth.so" modparam("auth", "nonce_expire", 300) modparam("auth", "rpid_suffix", ";party=calling;id-type=subscriber;screen=yes") modparam("auth", "rpid_avp", "$avp(s:rpid)") loadmodule "auth_db.so" modparam("auth_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "user_column", "username") modparam("auth_db", "domain_column", "domain") modparam("auth_db", "password_column", "password") modparam("auth_db", "password_column_2", "ha1b") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "use_domain", 0) modparam("auth_db", "load_credentials", "rpid") loadmodule "uri_db.so" modparam("uri_db", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("uri_db", "uri_table", "uri") modparam("uri_db", "uri_user_column", "username") modparam("uri_db", "uri_domain_column", "domain") modparam("uri_db", "uri_uriuser_column", "uri_user") modparam("uri_db", "subscriber_table", "subscriber") modparam("uri_db", "subscriber_user_column", "username") modparam("uri_db", "subscriber_domain_column", "domain") modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "use_domain", 0) loadmodule "lcr.so" modparam("lcr", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("lcr", "gw_table", "gw") modparam("lcr", "gw_name_column", "gw_name") modparam("lcr", "ip_addr_column", "ip_addr") modparam("lcr", "port_column", "port") modparam("lcr", "uri_scheme_column", "uri_scheme") modparam("lcr", "transport_column", "transport") modparam("lcr", "grp_id_column", "grp_id") modparam("lcr", "lcr_table", "lcr") modparam("lcr", "strip_column", "strip") modparam("lcr", "prefix_column", "prefix") modparam("lcr", "from_uri_column", "from_uri") modparam("lcr", "priority_column", "priority") modparam("lcr", "gw_uri_avp", "1400") modparam("lcr", "ruri_user_avp", "1402") modparam("lcr", "contact_avp", "1401") modparam("lcr", "fr_inv_timer_avp", "s:fr_inv_timer_avp") modparam("lcr", "fr_inv_timer", 90) modparam("lcr", "fr_inv_timer_next", 30) modparam("lcr", "rpid_avp", "s:rpid") ######################################################################## # Request route 'main' ######################################################################## route[0] { xlog("L_INFO", "New request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); force_rport(); if(msg:len > max_len) { xlog("L_INFO", "Message too big - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("513", "Message Too Big"); exit; } if (!mf_process_maxfwd_header("10")) { xlog("L_INFO", "Too many hops - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("483", "Too Many Hops"); exit; } if(!is_method("REGISTER")) { if(nat_uac_test("19")) { record_route(";nat=yes"); } else { record_route(); } } if(is_method("CANCEL") || is_method("BYE")) { unforce_rtp_proxy(); } if(loose_route()) { if(!has_totag()) { xlog("L_INFO", "Initial loose-routing rejected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Initial Loose-Routing Rejected"); exit; } if(nat_uac_test("19") || search("^Route:.*;nat=yes")) { fix_nated_contact(); setbflag(6); } route(3); } if(is_method("REGISTER")) { route(2); } if(is_method("INVITE")) { route(4); } if(is_method("CANCEL") || is_method("ACK")) { route(8); } route(9); } ######################################################################## # Request route 'stop-rtp-proxy' ######################################################################## route[1] { if(isflagset(22)) { unforce_rtp_proxy(); } } ######################################################################## # Request route 'base-route-register' ######################################################################## route[2] { sl_send_reply("100", "Trying"); if(!www_authorize("", "subscriber")) { xlog("L_INFO", "Register authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); www_challenge("", "0"); exit; } if(!check_to()) { xlog("L_INFO", "Spoofed To-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed To-URI Detected"); exit; } consume_credentials(); if(!search("^Contact:[ ]*\*") && nat_uac_test("19")) { fix_nated_register(); setbflag(6); } if(!save("location")) { xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_reply_error(); exit; } xlog("L_INFO", "Registration successful - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); exit; } ######################################################################## # Request route 'base-outbound' ######################################################################## route[3] { if(isbflagset(6)) { if(!isflagset(22) && !search("^Content-Length:[ ]*0")) { setflag(22); force_rtp_proxy(); } t_on_reply("2"); } else { t_on_reply("1"); } if(!isflagset(21)) { t_on_failure("2"); } if(isflagset(29)) { append_branch(); } if(is_present_hf("Proxy-Authorization")) { consume_credentials(); } xlog("L_INFO", "Request leaving server, D-URI='$du' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # no 100 (we already sent it) and no DNS blacklisting if(!t_relay("0x05")) { sl_reply_error(); if(is_method("INVITE") && isbflagset(6)) { unforce_rtp_proxy(); } } exit; } ######################################################################## # Request route 'base-route-invite' ######################################################################## route[4] { sl_send_reply("100", "Trying"); if(from_gw()) { xlog("L_INFO", "Call from PSTN' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); setflag(23); } else { if(!proxy_authorize("", "subscriber")) { xlog("L_INFO", "Proxy authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); proxy_challenge("", "0"); exit; } if(!check_from()) { xlog("L_INFO", "Spoofed From-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed From-URI Detected"); exit; } } if(nat_uac_test("19")) { fix_nated_contact(); setbflag(6); } route(5); } ######################################################################## # Request route 'invite-find-callee' ######################################################################## route[5] { if(!is_domain_local("$rd")) { setflag(20); route(7); } if(does_uri_exist()) { xlog("L_INFO", "Callee is local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(6); } else { xlog("L_INFO", "Callee is not local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(7); } exit; } ######################################################################## # Request route 'invite-to-internal' ######################################################################## route[6] { if(!lookup("location")) { xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Offline"); } else { xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(3); } exit; } ######################################################################## # Request route 'invite-to-external' ######################################################################## route[7] { if(isflagset(20)) { xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(3); exit; } if(!isflagset(23)) { # don't allow calls relaying from PSTN to PSTN, if not explicitely forwarded if(uri =~ "^sip:[0-9]+@") { # only route numeric users to PSTN if(!load_gws()) { xlog("L_ERR", "Error loading PSTN gateways - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } if(!next_gw()) { xlog("L_ERR", "No PSTN gateways available - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } setflag(21); t_on_failure("1"); route(3); } } xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Not Found"); exit; } ######################################################################## # Request route 'base-route-local' ######################################################################## route[8] { t_on_reply("1"); if(t_check_trans()) { xlog("L_INFO", "Request leaving server - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(!t_relay()) { sl_reply_error(); } } else { xlog("L_INFO", "Dropping mis-routed request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } exit; } ######################################################################## # Request route 'base-route-generic' ######################################################################## route[9] { xlog("L_INFO", "Method not supported - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("501", "Method Not Supported Here"); exit; } ######################################################################## # Request route 'base-filter-failover' ######################################################################## route[10] { if(!t_check_status("408|500|503")) { xlog("L_INFO", "No failover routing needed for this response code - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(1); exit; } } ######################################################################## # Reply route 'base-standard-reply' ######################################################################## onreply_route[1] { xlog("L_INFO", "Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); exit; } ######################################################################## # Reply route 'base-nat-reply' ######################################################################## onreply_route[2] { xlog("L_INFO", "NAT-Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); if(nat_uac_test("1")) { fix_nated_contact(); } if(isbflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if(!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); } } exit; } ######################################################################## # Failure route 'pstn-failover' ######################################################################## failure_route[1] { xlog("L_INFO", "Failure route for PSTN entered - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(10); if(!next_gw()) { xlog("L_ERR", "Failed to select next PSTN gateway - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(1); exit; } t_on_failure("1"); route(3); } ######################################################################## # Failure route 'base-standard-failure' ######################################################################## failure_route[2] { route(10); route(1); }

2 1

[OpenSER-Users] Missed calls
by David Villasmil Govea 14 Dec '07

14 Dec '07

Hello all, I've configured Openser 1.2 with accounting + mysql. at the momento i'm only loggin: BYE CANCEL INVITE REGISTER * How do i identify missed calls? * Furthermore, Openser doesn't see all INVITES+BYE, so i can't make cdrs!! I've read somewhere that to log all transaction messages i should have "strict route" as oppossed to "loose route". Is that enabled by: modparam("rr", "enable_full_lr", 0) ?? Thanks a lot! this is my config: # # sample config file to be used with nathelper/rtpproxy # # start RTPProxy with: rtpproxy -l your_public_ip -s udp:localhost:8899 # # ----------- global configuration parameters ------------------------ debug=2 # 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=20 disable_dns_blacklist=true # # --- module loading mpath="/usr/lib/openser/modules/" loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "nathelper.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "mi_fifo.so" loadmodule "dispatcher.so" #loadmodule "domain.so" loadmodule "xlog.so" loadmodule "/usr/lib/openser/modules/acc.so" modparam("acc", "early_media", 1) modparam("acc", "report_cancels", 1) modparam("acc", "db_flag", 2) modparam("acc", "db_url", "mysql://user:password@localhost/openser") modparam("acc", "report_ack", 0) modparam("acc", "db_missed_flag", 3) modparam("acc", "failed_transaction_flag", 4) modparam("acc", "db_extra", "from_uri=$fu ; to_uri=$tu ; from_user=$fU; source_ip=$si") # # CONSTANTS # # --- setting module parameters #modparam("domain","db_url","mysql://user:password@localhost/openser") #modparam("domain", "db_mode", 1) #modparam("domain", "domain_table", "domain") #modparam("domain", "domain_col", "domain") modparam("xlog", "buf_size", 8192) # --- setting module parameters modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("usrloc|auth_db","db_url","mysql://user:password@localhost /openser") # -- usrloc params -- modparam("usrloc", "db_mode", 1) modparam("usrloc", "nat_bflag", 6) #modparam("usrloc", "use_domain", 0) # -- registrar params -- modparam("registrar|nathelper", "received_avp", "$avp(i:42)") # -- auth params -- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") #modparam("auth_db", "use_domain", 0) # -- rr params -- #modparam("rr", "enable_full_lr", 1) # -- nathelper modparam("nathelper", "rtpproxy_sock", "udp:1.2.3.5:8899") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@sip2.telmaxi.com.br") #modparam("registrar", "max_contacts", 1) # --- main routing logic route{ if (method=="INVITE") { xlog("L_NOTICE", "INVITE MESSAGE RECEIVED - START ACC\n"); setflag(2); setflag(3); setflag(4); }; if (method=="BYE") { xlog ("L_NOTICE", "BYE - STOP ACCOUNTING\n"); setflag(2); setflag(3); setflag(4); }; if (method=="CANCEL") { xlog ("L_NOTICE", "CANCEL - STOP ACCOUNTING\n"); setflag(2); setflag(3); setflag(4); }; if (method=="REGISTER") { setflag(2); setflag(3); setflag(4); }; if (!mf_process_maxfwd_header("100")) { sl_send_reply("483","Too Many Hops"); exit; }; #if (msg:len >= 2048 ) { # sl_send_reply("513", "Message too big"); # exit; #}; # NAT detection route(2); if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # if (!is_uri_host_local() ) { append_hf("P-hint: outbound\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER") { xlog("L_NOTICE", "\nNew Registration\n"); if (!www_authorize("1.2.3.4", "subscriber")) { xlog("L_NOTICE", "\nRegistration FAILED! CLIENT: $au\n"); www_challenge("1.2.3.4", "0"); exit; }; if (isflagset(5)) { xlog("L_NOTICE", "\nRegistration OK\n"); setbflag(6); # if you want OPTIONS natpings uncomment next # setbflag(7); }; save("location"); exit; }; if (uri=~"^sip:0"){ xlog("L_NOTICE", "\n************** GOT OUTSIDE CALL ************\n"); ds_select_dst("1","2"); sl_send_reply("100","Trying"); forward(); exit(); } if (uri=~"^sip:7"){ ds_select_dst("1","2"); sl_send_reply("100","Trying"); forward(); exit(); } if (uri=~"^sip:1"){ ds_select_dst("1","2"); sl_send_reply("100","Trying"); forward(); exit(); } if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); #}; }; route(1); } route[1] { xlog("L_NOTICE", "\n**************IN ROUTE 1************\n"); if (subst_uri('/(sip:.*);nat=yes/\1/')){ setbflag(6); }; if (isflagset(5)||isbflagset(6)) { route(3); } if (!t_relay()) { sl_reply_error(); }; exit; } route[2]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; setflag(5); }; } route[3] { if (is_method("BYE|CANCEL")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); t_on_failure("1"); }; if (isflagset(5)) search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); t_on_reply("1"); } failure_route[1] { if (isbflagset(6) || isflagset(5)) { unforce_rtp_proxy(); } } onreply_route[1] { if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); if (isbflagset(6)) { fix_nated_contact(); } exit; }

2 1

[OpenSER-Users] Security hole in REGISTER's Contact using domain
by Iñaki Baz Castillo 14 Dec '07

14 Dec '07

Hi, Permissions module tries to avoid REGISTER with privileged IP's in Contact (using "register.deny" file) but I have some doubs about this security. I'll play with the example explained in "register.deny" file: --------------------------------------------------------------------------------------- # Suppose that we have a PSTN gateway with IP address 1.2.3.4 # We should prevent REGISTER messages that contain that IP # address in Contact header field because that can cause serious # security hole (a malicious user might be able to register such # a contact and bypass security checks performed by the SIP proxy). # # The following line prevents registering Contacts with IP 1.2.3.4 # (Don't forget to list also all hostnames that can be used to # reach the PSTN gateway) ALL : "^sip:.*1\.2\.3\.4" --------------------------------------------------------------------------------------- Ok, now a malicious user could just use SipSak to send a malicious REGISTER to call for free to a PSTN number 01666555444: ~# sipsak -U -C sip:01666555444@1.2.3.00004 -a passwd -s sip:200@domain.org Note the "000004" !!!! So this causes a entry in "location" with fields: - username = 200 - domain = domain.org - contact = sip:01666555444@1.2.3.00004 And sure 1.2.3.00004 is a valid IPv4. This is: if the user calls itself (sip:200@domain.org) he'll get a free PSTN call. Oppss... Ok, a solution could be to improve the regular expression by avoiding any number of 0's: ALL : "^sip:.*0*1\.0*2\.0*3\.0*4" Ok, but now the malicious user can register a domain "hacking_my_proxy.com" to resolve to IP 1.2.3.4, and send this REGISTER: ~# sipsak -U -C sip:01666555444@hacking_my_proxy.com -a passwd -s sip:200@domain.org So this will bypass the "register.deny" policy !!!! Note that "register.deny" file says: # (Don't forget to list also all hostnames that can be used to # reach the PSTN gateway) Of course, it's not possible to list all hostnames and domain resolving an IP (anyone can register a domain to any IP). So then... is it really valid this "register.deny" security???? Solution for this? ------------------------- - Forbid hostnames or domains in Contact: Ohh, too much anti-RFC 3261 (what would "alice(a)pc33.atlanta.com" think about it? XDDD). - Do a DNS query for the "Contact" during REGISTER: What about if DNS changes later? - Match the resolved IP against IP's in "register.deny" for every INVITE leaving OpenSer. Humm. - Avoid OpenSer using internet DNS system (so "hacking_my_proxy.com" wouldn't be resolved) and allow just secure domains (internal DNS or /etc/hosts): and what about outbound calls? isn't this solution an atrocity? How to handle it? is it not a real security hole? Comments are welcome. Regards. -- Iñaki Baz Castillo

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[OpenSER-Users] which media server to use to write a Ringback Tone application
by Srinivas Kotamarti 14 Dec '07

14 Dec '07

All, I am trying to write a carrier grade Ringback Tone application on top of OpenSER, does anybody know which media server (preferably OpenSource) to use? Is WeSIP the only way to write an application on top of OpenSER or there are otherways? Also, does anybody know how the applications in WeSIP perform can they be deployed as a carrier grade solution? Regards --Srinivas Kotamarti

2 1

[OpenSER-Users] 1.3.0 FreeBSD port committed to svn
by Jesus Rodriguez 14 Dec '07

14 Dec '07

Hi, Yesterday night i committed to the svn the update of the FreeBSD port to 1.3.0 version. This update will also be committed in the FreeBSD ports tree as soon as possible. Meanwhile, if you want to use the port or create your own packages, you can obtain the port here: http://openser.svn.sourceforge.net/viewvc/openser/branches/1.3/packaging/fr… Regards. Saludos JesusR. ------------------------------------ Jesus Rodriguez VozTelecom Sistemas, S.L. jesusr(a)voztele.com http://www.voztele.com Tel. 902360305 -------------------------------------

1 0

[OpenSER-Users] PARALLEL FORKING
by Gomtesh Jain 14 Dec '07

14 Dec '07

Hi All , I am trying to do parallel forking from Openser.But when I call then one user gets 2 calls and rest of the users get one call . Please any one explain why it is happening. Here I attach my openser.cfg file... # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls: domain2.net"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication #if (!www_authorize("192.168.3.15", "subscriber")) { # www_challenge("192.168.3.15", "0"); # exit; #}; save("location"); exit; }; lookup("aliases"); log("I M IN lookup"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; } else { append_branch(); append_hf("P-hint: usrloc applied\r\n"); t_relay(); } }; route(1); }

2 1

Re: [OpenSER-Users] Preparing the release of OpenSER 1.3 "BlackCat"
by Sergio Gutierrez 14 Dec '07

14 Dec '07

Hi Bogdan. The compilation on Solaris 10/SPARC has ended succesfully. Acording to my current setup, in particular, related to fulfilled dependencies, my final gmake command was: *gmake exclude_modules="jabber mi_xmlrpc osp perl postgres unixodbc snmpstats xmpp tlsops db_berkeley ldap mi_dm perlvdb xcap_client" *I have mysql 5.0.22 and radiusclient 0.5.2 The attached file contains the whole warnings reported at the compilation process; most of them are related to assignments without casting, and implicit declaration of functions. I will peform the TLS compilation, and the testing on Solaris x86, and I will be notifying you as soon as they end. If there is something I can help, or another test you would like I perform, please let me know. Best Regards. Sergio Gutiérrez EPM Telecomunicaciones Medellin, Colombia Tel: +574 - 3802947 On Dec 12, 2007 4:54 PM, Sergio Gutierrez <saguti(a)gmail.com> wrote: > Hi Bogdan. > > Continuing with compilation, this error appeared when compiling LDAP > module: > > In file included from ldap_api_fn.c:40: > ldap_api_fn.h:57: error: conflicting types for 'ldap_url_search' > /usr/include/ldap.h:789: error: previous declaration of 'ldap_url_search' > was here > ldap_api_fn.h:57: error: conflicting types for 'ldap_url_search' > /usr/include/ldap.h:789: error: previous declaration of 'ldap_url_search' > was here > ldap_api_fn.c: In function `ldap_params_search': > ldap_api_fn.c:211: warning: implicit declaration of function > `LDAP_API_ERROR' > > In my case, I do not have OpenLDAP libraries installed, but my system has > the LDAP library which is standard on Solaris since Solaris 9. > > checking at the manual, I see the declaration on Solaris is: > > int *ldap_url_search*(LDAP *ld, char *url, int attrsonly); > > > Please, let me know how can we continue. > > Regards. > > Sergio. > > > > On Dec 12, 2007 4:22 PM, Bogdan-Andrei Iancu < bogdan(a)voice-system.ro> > wrote: > > > Hi Sergio, > > > > Thanks for the feedback - I'm glad we were able to catch this issue in > > last minutes before release :). I also applied your patch for the > > Makefile of the carrierroute module. > > > > Regards, > > Bogdan > > > > Sergio Gutierrez wrote: > > > Hi Bogdan. > > > > > > The compilation so far is going right, with no warnings; > > > > > > A modification I did is I installed libconfuse, because for our > > > particular setup we are interested in exploring the carrierroute > > > module; I installed at my LOCALBASE, and at compiling the module, it > > > failed; trying to fix it I added a couple lines to the Makefile of the > > > module as indicated at the following diff; I am not sure if they are > > > required, but they fixed my problem: > > > > > > > > > ------------------------- BEGIN DIFF > > > --- Makefile Wed Dec 12 16:05:06 2007 > > > +++ Makefile.saguti Wed Dec 12 16:04:58 2007 > > > @@ -8,7 +8,8 @@ > > > include ../../Makefile.defs > > > auto_gen= > > > NAME=carrierroute.so > > > -LIBS=-lconfuse > > > +LIBS+=-L$(LOCALBASE)/lib -lconfuse > > > +DEFS+=-I$(LOCALBASE)/include > > > > > > include ../../Makefile.modules > > > > > > ------------------------ END DIFF > > > > > > I will try again the compilation, and I hope to test it too on Solaris > > > x86. > > > > > > Best regards. > > > > > > Sergio. > > > > > > > >

2 4

[OpenSER-Users] OpenSER 1.3.0 BlackCat is released !
by Bogdan-Andrei Iancu 14 Dec '07

14 Dec '07

Hi, I'm glad to announce that OpenSER 1.3.0 was released - I want to thanks to all people that contributed to it, as code, documentation, assistance, packaging, etc... The SVN branch for 1.3 and tag for 1.3.0 was created, so feel free to download : # svn co https://openser.svn.sourceforge.net/svnroot/openser/branches/1.3 openser As tarballs, only source (tls and non-tls) and linux binaries are available on openser.org site. In the next following days packages will be uploaded on both openser.org and SF sites. There are still pending to upload on the sites a news about the features in 1.3.0, about migration from 1.2 to 1.3 - all these will be done in the following days. More announcement email (for detailed) will follow ;). Regards, Bogdan

4 5

[OpenSER-Users] SUSE packages
by Daniel-Constantin Mierla 13 Dec '07

13 Dec '07

Thanks to Peter Nixon, the OpenSER 1.3.0 packages for different flavors of SUSE are already available: http://download.opensuse.org/repositories/network:/telephony/ More details about how to use the repository with different package managers: http://www.openser.org/mos/view/Download/ Cheers, Daniel

1 0

[OpenSER-Users] openser 1.3.0 release notes
by Daniel-Constantin Mierla 13 Dec '07

13 Dec '07

Hello, the release notes for version 1.3.0 are available at: http://www.openser.org/mos/view/OpenSER-v1.3.x-Release-Notes/ If you feel that something is missing, let me know to update the page. Enjoy the new version! Daniel

1 0

[Serusers] loose_ route domain import
by Jan Ruzicka 13 Dec '07

13 Dec '07

Hi, does loose_route import the domains from domain module? My multidomain 0.9.6 seems no to do this and I found same issues for openser (http://www.mail-archive.com/users@openser.org/msg12794.html). What about 2.x? I'm afraid the only workaround, as mentioned, is to use alias=domain for all domains served by server. Thanks Jan -- ------------------------------------------------------------------- Ing. Jan Ruzicka CESNET z. s. p. o. tel: +420234680499 Zikova 4 e-mail,sip: janru(a)cesnet.cz 160 00 Praha 6 icq:151316768 Czech Republic -------------------------------------------------------------------

5 4

Re: [OpenSER-Users] OpenSER as SIP Proxy for PSTN Gateway
by Andy Smith 13 Dec '07

13 Dec '07

>Yes, define a peer in sip.conf: > >[openser] >host=XX.XX.XX.XX >insecure=invite > >In this way Asterisk allow calls from OpenSer without forcing them to >authenticate. Authenticacion had already place in OpenSer.So I guess this would function for calls coming from the PSTN gateway, what about for callsoriginating from SIP devices connected to Asterisk? Can I do some similar config on OpenSERso that calls authenticated on Asterisk dont require authentication for an INVITE on OpenSER?

2 1

[Serusers] SER authentication through FreeRadius with an LDAPbackend
by Daniel Serrano 13 Dec '07

13 Dec '07

Dear Greger V. Teigre, Did you find another solution for this issue instead of generate the new attribute in the LDAP DB? Some kind of workaround will be great!.. Thanks in advance. Regards, Daniel Serrano.

2 4

[Serusers] PARALLEL FORKING
by Gomtesh Jain 13 Dec '07

13 Dec '07

Hi All , I am trying to do parallel forking from SER.But when I call then one user gets 2 calls and rest of the users get one call . Please any one explain why it is happening. Here I attach my openser.cfg file... # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication #if (!www_authorize("192.168.3.15", "subscriber")) { # www_challenge("192.168.3.15", "0"); # exit; #}; save("location"); exit; }; lookup("aliases"); log("I M IN lookup"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; } else { append_branch(); append_hf("P-hint: usrloc applied\r\n"); t_relay(); } }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP #if (!t_relay()) { if (!forward()) { sl_reply_error(); }; exit; } Gomtesh

2 1

[Serusers] Can SER be used as a SIP A/S
by Srinivas Kotamarti 13 Dec '07

13 Dec '07

All, Can SER be used as a SIP A/S for example to write a carrier grade Ringback Tone application on top of SER. Does anybody know which media server (preferably OpenSource) to use? Is SEMS the only way to write an application on top of SER or there are other ways? Also, does anybody know how the applications in SEMS perform can they be deployed as a carrier grade solution? Regards --Srinivas Kotamarti

1 0

Re: [OpenSER-Users] OpenSER as SIP Proxy for PSTN Gateway
by Andy Smith 13 Dec '07

13 Dec '07

>Register is used JUST to **RECEIVE** calls, not to send calls. > >SIP uses http-digest autentication. This means: in **EVERY** INVITE a proxy >wnat to authenticate the client (Asterisk) receives a challenge from the >proxy and the client must generate a hash (a response) with its credentials >and the nonce received.Ok, thanks for the reply Iñaki. So in the case of a configuration like mine what are other people doing to negate theneed to define all the users in both Asterisk and OpenSER? Are you authenticating the users on OpenSER basedon source IP? Or allowing all connections without authentication? And in general any pointers on the best way for me to achieve a simple accountig solution that is easy to maintain as more asterisk boxes and users are added?

2 1

Re: [Serusers] Can SER be used as a SIP A/S
by Jiri Kuthan 13 Dec '07

13 Dec '07

At 19:54 13/12/2007, Srinivas Kotamarti wrote: >Content-Language: en-US >Content-Type: multipart/alternative; > boundary="_000_95BADEF3CDEB8441838C6AD39FA619A31343E206C0NAMAEMAILnmss_" > >All, >Can SER be used as a SIP A/S for example to write a carrier grade Ringback Tone application on top of SER. > >Does anybody know which media server (preferably OpenSource) to use? > >Is SEMS the only way to write an application on top of SER or there are other ways? Use SEMS. > >Also, does anybody know how the applications in SEMS perform can they be deployed as a carrier grade solution? Very well. -jiri > > >Regards > >--Srinivas Kotamarti > >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

1 0

[OpenSER-Users] Openser + radiusclient accounting need more information
by Fernando Yubero 13 Dec '07

13 Dec '07

Hi group, Im new in the mailing list, and starting again with Openser, last time I used was first release like an alternative to SER. Later dont run any SIP service. Actually Im testing OpenSer for a new service, voice service with PSTN, etc, and Im testing accounting via radius. I use this help to work basic things. www.openser.org/docs/openser-radius-1.0.x.html thanks for this documentation very helpful for me. But I dont get same result. My version Openser 1.2.2. I get this fields in start and stop, very less than example. Wed Dec 12 17:33:12 2007 Acct-Status-Type = Start Service-Type = SIP Sip-Response-Code = 200 Sip-Method = 1 Event-Timestamp = "Dec 12 2007 17:33:05 ART" Sip-From-Tag = "94ab8960-c82d7718-13c4-7db17-4cc178eb-7db17" Sip-To-Tag = "94b908a0-c82d7717-13c4-3815-ecb28f3-3815" Acct-Session-Id = "94ab6a20-c82d7718-13c4-7db17-7fcaa8b-7db17(a)elmejorcable.com.ar" NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = xxx.xxx.xxx.3 Client-IP-Address = xxx.xxx.xxx.3 Acct-Unique-Session-Id = "49ff3575432365d6" Timestamp = 1197491592 Wed Dec 12 17:33:19 2007 Acct-Status-Type = Stop Service-Type = SIP Sip-Response-Code = 200 Sip-Method = Bye Event-Timestamp = "Dec 12 2007 17:33:11 ART" Sip-From-Tag = "94b908a0-c82d7717-13c4-3815-ecb28f3-3815" Sip-To-Tag = "94ab8960-c82d7718-13c4-7db17-4cc178eb-7db17" Acct-Session-Id = "94ab6a20-c82d7718-13c4-7db17-7fcaa8b-7db17(a)elmejorcable.com.ar" NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = xxx.xxx.xxx.3 Client-IP-Address = xxx.xxx.xxx.3 Acct-Unique-Session-Id = "49ff3575432365d6" Timestamp = 1197491599 I think is about default fields to send for accounting via radius, How to get full info??? With radius_extra modparam??? Thanks in advanced. Fernando Yubero Communication Partner

1 0

[OpenSER-Users] Preparing the release of OpenSER 1.3 "BlackCat"
by Bogdan-Andrei Iancu 13 Dec '07

13 Dec '07

Hi, As scheduled, tomorrow will be the official release of OpenSER version 1.3. NAME ===== Following the proposal and the feedback from the lists, we will start to label each major version number with a name - it is more friendly . Note that the version numbering will still be the primary ID. OpenSER 1.3 will be named "BlackCat". CODE ===== As far as I checked the lists and tracker, all the critical and important bugs were fixed. No pending issues, as far as I know. Tomorrow I will run some compiling tests on several arch and OS to see if any warning is reported. PS: does anybody offer some access on SUN+SOLARIS ? As there are only a small number of pending changes on SVN (warnings and docs), probably in the afternoon, the 1.3 branch will be created. DOCS ===== The module documentation is up to date, thanks to the policy of synchronization between the code and doc updates. Just in case, I will re-generate all the text READMEs from the SGML files. Thanks to Henning, the changelog and Credits files were updated on SVN. Thanks to Daniel, we have a start for the the "New Features" document (see http://www.openser.org/dokuwiki/doku.php/sandbox:new-in-1.3.x). The idea will be to have it finalized y tomorrow and imported in the web pages (openser.org and SF). There will be a need for a overview of the "INSTALL" helper in order to be updated for 1.3 (maybe the DB part only). I suggest to put together all the docs from openser and upload them also on SF - of course this is not a priority, it can be done later, in the following days. TARBALLS ======== Tarballs with sources and binaries (linux i386) will be uploaded on openser.org and SF sites. Of course after the creation of the new 1.3 branch on SVN. Any other contributions (binaries of other OSs or archs) are welcomed - just send an email to team(a)openser.org . PACKAGES ======== After the official release, specs will be updated (if not yet) and packages will be generated. Please help with this process and let us know if you can generate any packages. Again use the lists or team(a)openser.org emails. I will generate the debian packages (unstable, stable, testing). The stable package will be also pushed on the openser.org APT repo. SITES UPDATES =========== Both openser.org and SF sites need to be updated for with news, docs, downloads, etc. This probably will take couple of days. Please let me know if there is something missing in the above list. Otherwise, if people are able and willing with some of the tasks, again, just let me know. Otherwise, have the champaign on cold for tomorrow ;)... Regards, Bogdan

2 4

Re: [OpenSER-Users] Could someone help me with WeSIP ?
by Daniel Grotti 13 Dec '07

13 Dec '07

Hi Elias, This is the WeSIP log's file, when I run startup.sh: 2007-12-12 10:02:57 HttpConnector Opening server socket on host IP address PROXY_IP 2007-12-12 10:02:57 Starting service Tomcat-Standalone 2007-12-12 10:02:57 HttpConnector[8080] Starting background thread 2007-12-12 10:02:57 HttpProcessor[8080][0] Starting background thread 2007-12-12 10:02:57 HttpProcessor[8080][1] Starting background thread 2007-12-12 10:02:57 HttpProcessor[8080][2] Starting background thread 2007-12-12 10:02:57 HttpProcessor[8080][3] Starting background thread 2007-12-12 10:02:57 HttpProcessor[8080][4] Starting background thread 2007-12-12 10:02:57 ConvergedEngine[Standalone]: Converged Application Server SIP&HTTP-0.1 2007-12-12 10:02:57 HostConfig[PROXY_IP]: Deploying configuration descriptor manager.xml 2007-12-12 10:02:57 WebappLoader[/manager]: Deploying class repositories to work directory /usr/src/wesip/work/Standalone/PROXY_IP/manager 2007-12-12 10:02:57 WebappLoader[/manager]: Deploy JAR /WEB-INF/lib/dwr.jar to /usr/src/wesip/wesipapps/../server/webapps/manager/WEB-INF/lib/dwr.jar 2007-12-12 10:02:57 WebappLoader[/manager]: Deploy JAR /WEB-INF/lib/jrobin-1.4.0.jar to /usr/src/wesip/wesipapps/../server/webapps/manager/WEB-INF/lib/jrobin-1.4.0.jar 2007-12-12 10:02:57 WebappLoader[/manager]: Deploy JAR /WEB-INF/lib/statistics.jar to /usr/src/wesip/wesipapps/../server/webapps/manager/WEB-INF/lib/statistics.jar 2007-12-12 10:02:58 ContextConfig[/manager]: Missing application sip.xml, using defaults only 2007-12-12 10:02:58 ContextConfig[/manager]: Configured an authenticator for method BASIC 2007-12-12 10:02:58 StandardManager[/manager]: Seeding random number generator class java.security.SecureRandom 2007-12-12 10:02:58 StandardManager[/manager]: Seeding of random number generator has been completed 2007-12-12 10:02:58 StandardHttpWrapper[/manager:default]: Loading container servlet default 2007-12-12 10:02:58 default: init 2007-12-12 10:02:58 dwr-invoker: init 2007-12-12 10:02:58 StandardHttpWrapper[/manager:invoker]: Loading container servlet invoker 2007-12-12 10:02:58 invoker: init 2007-12-12 10:02:58 jsp: init 2007-12-12 10:02:58 statistics: init 2007-12-12 10:02:58 StandardHttpContext[/manager]: StandardHttpContext [WeSIP Manager Application] has been started 2007-12-12 10:02:58 HostConfig[PROXY_IP]: Expanding web application archive click2call.sar 2007-12-12 10:02:59 ManagerBase[/click2call]: Seeding random number generator class java.security.SecureRandom 2007-12-12 10:02:59 ManagerBase[/click2call]: Seeding of random number generator has been completed 2007-12-12 10:02:59 ManagerBase[/click2call]: Seeding random number generator class java.security.SecureRandom 2007-12-12 10:02:59 ManagerBase[/click2call]: Seeding of random number generator has been completed 2007-12-12 10:02:59 WebappLoader[/click2call]: Deploying class repositories to work directory /usr/src/wesip/work/Standalone/PROXY_IP/click2call 2007-12-12 10:02:59 WebappLoader[/click2call]: Deploy class files /WEB-INF/classes to /usr/src/wesip/wesipapps/click2call/WEB-INF/classes 2007-12-12 10:02:59 WebappLoader[/click2call]: Deploy JAR /WEB-INF/lib/activation.jar to /usr/src/wesip/wesipapps/click2call/WEB-INF/lib/activation.jar 2007-12-12 10:02:59 WebappLoader[/click2call]: Deploy JAR /WEB-INF/lib/axis-ant.jar to /usr/src/wesip/wesipapps/click2call/WEB-INF/lib/axis-ant.jar 2007-12-12 10:02:59 WebappLoader[/click2call]: Deploy JAR /WEB-INF/lib/axis.jar to /usr/src/wesip/wesipapps/click2call/WEB-INF/lib/axis.jar 2007-12-12 10:02:59 WebappLoader[/click2call]: Deploy JAR /WEB-INF/lib/commons-discovery-0.2.jar to /usr/src/wesip/wesipapps/click2call/WEB-INF/lib/commons-discovery-0.2.jar 2007-12-12 10:02:59 WebappLoader[/click2call]: Deploy JAR /WEB-INF/lib/commons-logging.jar to /usr/src/wesip/wesipapps/click2call/WEB-INF/lib/commons-logging.jar 2007-12-12 10:02:59 WebappLoader[/click2call]: Deploy JAR /WEB-INF/lib/dwr.jar to /usr/src/wesip/wesipapps/click2call/WEB-INF/lib/dwr.jar 2007-12-12 10:02:59 WebappLoader[/click2call]: Deploy JAR /WEB-INF/lib/jaxrpc.jar to /usr/src/wesip/wesipapps/click2call/WEB-INF/lib/jaxrpc.jar 2007-12-12 10:02:59 WebappLoader[/click2call]: Deploy JAR /WEB-INF/lib/log4j-1.2.8.jar to /usr/src/wesip/wesipapps/click2call/WEB-INF/lib/log4j-1.2.8.jar 2007-12-12 10:02:59 WebappLoader[/click2call]: Deploy JAR /WEB-INF/lib/mail.jar to /usr/src/wesip/wesipapps/click2call/WEB-INF/lib/mail.jar 2007-12-12 10:02:59 WebappLoader[/click2call]: Deploy JAR /WEB-INF/lib/saaj.jar to /usr/src/wesip/wesipapps/click2call/WEB-INF/lib/saaj.jar 2007-12-12 10:02:59 WebappLoader[/click2call]: Deploy JAR /WEB-INF/lib/wsdl4j-1.5.1.jar to /usr/src/wesip/wesipapps/click2call/WEB-INF/lib/wsdl4j-1.5.1.jar 2007-12-12 10:02:59 StandardManager[/click2call]: Seeding random number generator class java.security.SecureRandom 2007-12-12 10:02:59 StandardManager[/click2call]: Seeding of random number generator has been completed 2007-12-12 10:02:59 StandardHttpWrapper[/click2call:default]: Loading container servlet default 2007-12-12 10:02:59 default: init 2007-12-12 10:02:59 init 2007-12-12 10:02:59 dwr-invoker: init 2007-12-12 10:02:59 init 2007-12-12 10:02:59 AxisServlet: init 2007-12-12 10:03:00 StandardHttpWrapper[/click2call:invoker]: Loading container servlet invoker 2007-12-12 10:03:00 invoker: init 2007-12-12 10:03:00 jsp: init 2007-12-12 10:03:00 AdminServlet: init 2007-12-12 10:03:00 SOAPMonitorService: init 2007-12-12 10:03:00 ConvergedContext[/click2call]: ConvergedContext [Click2Call SipServlet] has been started 2007-12-12 10:03:00 HostConfig[PROXY_IP]: Deploying web application directory click2call-08_11_06 2007-12-12 10:03:00 ManagerBase[/click2call-08_11_06]: Seeding random number generator class java.security.SecureRandom 2007-12-12 10:03:00 ManagerBase[/click2call-08_11_06]: Seeding of random number generator has been completed 2007-12-12 10:03:00 ManagerBase[/click2call-08_11_06]: Seeding random number generator class java.security.SecureRandom 2007-12-12 10:03:00 ManagerBase[/click2call-08_11_06]: Seeding of random number generator has been completed 2007-12-12 10:03:00 WebappLoader[/click2call-08_11_06]: Deploying class repositories to work directory /usr/src/wesip/work/Standalone/PROXY_IP/click2call-08_11_06 2007-12-12 10:03:00 WebappLoader[/click2call-08_11_06]: Deploy class files /WEB-INF/classes to /usr/src/wesip/wesipapps/click2call-08_11_06/WEB-INF/classes 2007-12-12 10:03:00 WebappLoader[/click2call-08_11_06]: Deploy JAR /WEB-INF/lib/activation.jar to /usr/src/wesip/wesipapps/click2call-08_11_06/WEB-INF/lib/activation.jar 2007-12-12 10:03:00 WebappLoader[/click2call-08_11_06]: Deploy JAR /WEB-INF/lib/axis-ant.jar to /usr/src/wesip/wesipapps/click2call-08_11_06/WEB-INF/lib/axis-ant.jar 2007-12-12 10:03:00 WebappLoader[/click2call-08_11_06]: Deploy JAR /WEB-INF/lib/axis.jar to /usr/src/wesip/wesipapps/click2call-08_11_06/WEB-INF/lib/axis.jar 2007-12-12 10:03:00 WebappLoader[/click2call-08_11_06]: Deploy JAR /WEB-INF/lib/commons-discovery-0.2.jar to /usr/src/wesip/wesipapps/click2call-08_11_06/WEB-INF/lib/commons-discovery-0.2.jar 2007-12-12 10:03:00 WebappLoader[/click2call-08_11_06]: Deploy JAR /WEB-INF/lib/commons-logging.jar to /usr/src/wesip/wesipapps/click2call-08_11_06/WEB-INF/lib/commons-logging.jar 2007-12-12 10:03:00 WebappLoader[/click2call-08_11_06]: Deploy JAR /WEB-INF/lib/dwr.jar to /usr/src/wesip/wesipapps/click2call-08_11_06/WEB-INF/lib/dwr.jar 2007-12-12 10:03:00 WebappLoader[/click2call-08_11_06]: Deploy JAR /WEB-INF/lib/jaxrpc.jar to /usr/src/wesip/wesipapps/click2call-08_11_06/WEB-INF/lib/jaxrpc.jar 2007-12-12 10:03:00 WebappLoader[/click2call-08_11_06]: Deploy JAR /WEB-INF/lib/log4j-1.2.8.jar to /usr/src/wesip/wesipapps/click2call-08_11_06/WEB-INF/lib/log4j-1.2.8.jar 2007-12-12 10:03:00 WebappLoader[/click2call-08_11_06]: Deploy JAR /WEB-INF/lib/mail.jar to /usr/src/wesip/wesipapps/click2call-08_11_06/WEB-INF/lib/mail.jar 2007-12-12 10:03:00 WebappLoader[/click2call-08_11_06]: Deploy JAR /WEB-INF/lib/saaj.jar to /usr/src/wesip/wesipapps/click2call-08_11_06/WEB-INF/lib/saaj.jar 2007-12-12 10:03:00 WebappLoader[/click2call-08_11_06]: Deploy JAR /WEB-INF/lib/wsdl4j-1.5.1.jar to /usr/src/wesip/wesipapps/click2call-08_11_06/WEB-INF/lib/wsdl4j-1.5.1.jar 2007-12-12 10:03:01 StandardManager[/click2call-08_11_06]: Seeding random number generator class java.security.SecureRandom 2007-12-12 10:03:01 StandardManager[/click2call-08_11_06]: Seeding of random number generator has been completed 2007-12-12 10:03:01 StandardHttpWrapper[/click2call-08_11_06:default]: Loading container servlet default 2007-12-12 10:03:01 default: init 2007-12-12 10:03:01 init 2007-12-12 10:03:01 dwr-invoker: init 2007-12-12 10:03:01 init 2007-12-12 10:03:01 AxisServlet: init 2007-12-12 10:03:01 StandardHttpWrapper[/click2call-08_11_06:invoker]: Loading container servlet invoker 2007-12-12 10:03:01 invoker: init 2007-12-12 10:03:01 jsp: init 2007-12-12 10:03:01 AdminServlet: init 2007-12-12 10:03:02 SOAPMonitorService: init 2007-12-12 10:03:02 ConvergedContext[/click2call-08_11_06]: ConvergedContext [Click2Call SipServlet] has been started 2007-12-12 10:03:02 HostConfig[PROXY_IP]: Deploying web application directory ROOT 2007-12-12 10:03:02 ManagerBase[]: Seeding random number generator class java.security.SecureRandom 2007-12-12 10:03:02 ManagerBase[]: Seeding of random number generator has been completed 2007-12-12 10:03:02 ManagerBase[]: Seeding random number generator class java.security.SecureRandom 2007-12-12 10:03:02 ManagerBase[]: Seeding of random number generator has been completed 2007-12-12 10:03:02 WebappLoader[]: Deploying class repositories to work directory /usr/src/wesip/work/Standalone/PROXY_IP/_ 2007-12-12 10:03:02 ContextConfig[]: Missing application sip.xml, using defaults only 2007-12-12 10:03:02 StandardManager[]: Seeding random number generator class java.security.SecureRandom 2007-12-12 10:03:02 StandardManager[]: Seeding of random number generator has been completed 2007-12-12 10:03:02 StandardHttpWrapper[:default]: Loading container servlet default 2007-12-12 10:03:02 default: init 2007-12-12 10:03:02 StandardHttpWrapper[:invoker]: Loading container servlet invoker 2007-12-12 10:03:02 invoker: init 2007-12-12 10:03:02 jsp: init 2007-12-12 10:03:02 ConvergedContext[]: ConvergedContext [Welcome] has been started 2007-12-12 10:03:02 > Default Wesip Application ==> http://PROXY_IP:8080/ > 2007-12-12 10:03:02 > In order to access the manager application use default credentials: - username: wesip - password: wesip > This configuration is defined in wesip-users.xml inside conf folder: - file: ./conf/wesip-users.xml I'd like to use " Click to dial " service. I don't write configuration file yet (sip.xml or web.xml). At this point, What should I do? Shouldn't I to access to Manager Wesip Application? (Manager Wesip Application ==> http://PROXY_IP:8080/manager/html?) - If yes, in this case I get an 500 http Error. My problem is very simple : I'm new in using servlet so I 'd like to know how I have to move (I didn't find no useful guide on this) . thanks. daniel -----Messaggio originale----- Da: elias baixas [mailto:elias.baixas@voztele.com] Inviato: mer 12/12/2007 0.40 A: Daniel Grotti Cc: users(a)lists.openser.org Oggetto: Re: [OpenSER-Users] Could someone help me with WeSIP ? Hello Daniel, can you provide some WeSIP's logs and openser's logs to gain some more information on the problem ? it's hard to guess what might be wrong with just the configuration. of course, you can substitute your IP addresses with some predefined token for the sake of confidentiality. for example: bash#>sed -i -e "s/192.168.1.1/PRIVATE_PROXY_IP/g" openser.cfg thanks !

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[OpenSER-Users] Any toturial or tip for using Seas module and wesip Ringtone application
by CHEN XUEQIN 13 Dec '07

13 Dec '07

Hi: I just installed wesip application server and 1.3.0-pre1-notls with seas.so modules. I also downloaded ringtone-24_05_07.sar application into path $WESIP_ROOT/wesipapps. Ijust configed openser.cfg and server.xml especially for those key property such as com.voztele.javax.sip.SER_ADDRESS, listen_sockets. The openser.cfg was mainly copied from http://www.openser.org/docs/modules/1.3.x standalone script. It seemed that openser and wesip server started correctly. Then I have the following test: 1. UA1 register to openser 2. UA2 register to openser 3. UA1 call UA2 through openser 4. UA1 ringing, received 100 trying, no 183 with SDP Can anyone tell me how to use ringtone applicaton? Any config should be done with ringtone application? Or my setup was wrong? Thanks in advance. Best Regards, Chen Xueqin

1 0

[OpenSER-Users] ORACLE-PORT
by Martin.Gruber1＠infineon.com 13 Dec '07

13 Dec '07

HI @ all, I have a short question: is it possible to add the Port to the Connectionstring of unixODBC in openser.cfg ? Thanks! Martin

2 1

[OpenSER-Users] migrating 1.2.x to 1.3.0
by Daniel-Constantin Mierla 13 Dec '07

13 Dec '07

Hello, those that experienced upgrades from 1.2.x to upcoming 1.3.0, may want to contribute to next wiki page, with found issues and their solutions: http://www.openser.org/dokuwiki/doku.php/install:1.2.2-to-1.3.0 Cheers, Daniel

1 0

[Serusers] acc module is not working wiht mysql module
by Rogelio Serrano 13 Dec '07

13 Dec '07

I think this is a bug. If the the accounting messages show up in syslog then it should also show up in ser.acc. -- Lay low and nourish in obscurity

1 0

[Serusers] mysql acc table empty
by Rogelio Serrano 13 Dec '07

13 Dec '07

I am totally stumped with the accounting module not inserting any data into the mysql "acc" table. I can see the accounting entries in syslog. There are no errors with mysql. I have started reading the source code as a last resort but i dont have enough time left. here is my config: debug=4 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) listen=sip.asianet.co.in /* 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/nathelper.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" loadmodule "/usr/local/lib/ser/modules/acc.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) modparam("usrloc", "db_url", "mysql://root:heslo@localhost/ser") # -- 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) # We will you flag 6 to mark NATed contacts modparam("registrar", "nat_flag", 6) # Enable NAT pinging modparam("nathelper", "natping_interval", 60) # Ping only contacts that are known to be # behind NAT modparam("nathelper", "ping_nated_only", 1) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 3) modparam("acc", "failed_transactions", 1) modparam("acc", "report_cancels", 1) modparam("acc", "db_url", "mysql://root:heslo@localhost/ser") # ------------------------- 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; }; # special handling for NATed clients; first, nat test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding used); also, # the received test should, if complete, should check all # vias for presence of received if (nat_uac_test("3")) { # allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart smart enough to be symmetric. In some phones, like # it takes a configuration option. With Cisco 7960, it is # called NAT_Enable=Yes, with kphone it is called # "symmetric media" and "symmetric signaling". (The latter # not part of public released yet.) fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP log("LOG: fixing nated sdp"); }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; # 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); t_relay(); break; }; if (method=="INVITE"){ setflag(1); } if (method=="BYE"){ setflag(1); } 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; }; #if (uri=~"^sip:1[0-9]*@") { # International PSTN # route(2); # break; #}; # native SIP destinations are handled using our USRLOC DB if (uri=~"^sip:8[0-9]*@") { # ip to ip if (lookup("location")) { append_hf("P-hint: usrloc applied\r\n"); route(1); break; } else { sl_send_reply("404", "Not Found"); break; }; } else { #send to pstn gateway route(2); }; }; } route[1] { # # Forcing media relay if necessary if (uri=~"[@:](192\.168\.|10\.|172\.16)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; if (isflagset(6)) { force_rtp_proxy(); t_on_reply("1"); append_hf("P-Behind-NAT: Yes\r\n"); }; # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- setflag(1); rewritehostport("192.168.1.1:5060"); # INSERT YOUR PSTN GATEWAY IP ADDRESS if (!t_relay()) { sl_reply_error(); }; } onreply_route[1] { if (status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); }; } -- Lay low and nourish in obscurity

1 0

[OpenSER-Users] WeSIP: Not SER connector available - Check the Configuration
by Franz Edler 13 Dec '07

13 Dec '07

Hi, I try to configure WeSIP which should not be so hard according to documentation. When I start OpenSER and WeSIP I get on the logs of OpenSER: -1(0) Fantastic, we have a new client: wesip1 -1(0) Fantastic, we have a new client: wesip1 -1(0) EUREKA, we have a new completed AS: wesip1 Also on WeSIP log I get: ExpressConnector [main] - SEAS module successfully connected at 192.168.0.5:5080 But when I call the manager Web-GUI I get as a status of OpenSER: Not SER connector available - Check the Configuration This seems to be in contradiction to above logs, isn't it? Any comment appreciated Franz

1 0

[OpenSER-Users] OpenSER as SIP Proxy for PSTN Gateway
by Andy Smith 12 Dec '07

12 Dec '07

Hi, As you may work out Im not an expert in this, hopefully for those of you that are this is a pretty straight forward question. Im trying to implement something like this: Cisco_as5300 <-----> OpenSER <------> Atersisk PSTN Gateway SIP Proxy PBX ^ Radius I am attempting to setup OpenSER as a SIP proxy for a Cisco PSTN Gateway, on the other side of OpenSER I have an Asterisk system. My confusion centres around the connection between the end users (SNOM or whatever) which hang off Asterisk and the connection between Asterisk and OpenSER. I have configure Asterisk to SIP register to OpenSER, all works ok here and Asterisk sees OpenSER as a SIP peer and channel. The bit Im not getting is that when someone dials a number from a handset connected to Asterisk that needs to route out via OpenSER, Asterisk seems to pass the username and password credentials of the handset to OpenSER. Firstly if its going to do this, then what is the point of SIP registering Asterisk as its authenticating ever session as individual entities anyway, secondly this isnt scaleable :( it seems to mean I have to configure each SNOM user both in Asterisk and in OpenSER. All of which doesnt seem very logical, can anyone tell me if I'm doing something wrong or if Im approaching this in the wrong way or if I need to approach the problem in some other way... The end goal of this is that Id like the ability to bill on a per SIP peer basis customer Asterisk boxes, I dont want to end up in a situation were I have to know about every SNOM a customer has. many many thanks in advance, Im a bit stuck at the mo! thanks Andy.

2 1

[Serusers] SER Incoming Calls
by Arun Kumar 12 Dec '07

12 Dec '07

HI User ----> Broadsoft -----> SER (Added Broadsoft ip as Trusted ) ------> Ser exten (Behind NAT) from SER exten I can dial my Broadsoft exten works fine. But reverse is not working fine. Incoming call from Broadsoft exten to my ser exten (027113) if (uri=~"^sip:027113@") { if (lookup("location")) { route(4); route(1); break; }; };

1 0

[OpenSER-Users] openser 1.3 packaging
by Daniel-Constantin Mierla 12 Dec '07

12 Dec '07

Hello, for those that want to start updating the spec files for packages and test them, I have created a README file that lists the new modules and their dependencies, as well as suggestions for where to place the modules. Also, I have uploaded today's source tarball, so all elements are in single place. Please check: http://www.openser.org/pub/openser/pre-1.3.0/ If you need further help, let me know. Comments and suggestions are welcome. Cheers, Daniel

4 3

[Serusers] SER Strange Calling Issue
by Arun Kumar 12 Dec '07

12 Dec '07

Hi User ---> SER -----> BroadoSoft -----> PSTN when I send calls to PSTN I'm changing the From to one of BroadSoft users b'coz without that BroadSoft don't allow me to Call PSTN. When call comes to PSTN Route I'm changing the the From : subst('/^From:(.*)sip:[^@]*@[a-zA-Z0-9.]+(.*)$/From:< sip:8233843435@Broadsoft IP>/ig'); calls goes fine to Broadsoft and then to PSTN but, once call connect to PSTN I'm able to hear other side but my phone always shows Calling and other side is not able to hear me (b'coz from my side channel is not getting answered). I tried forcing mediaproxy before PSTN Calls but no luck. Please advise. Its very urgent. Thanks, Arun

1 1

[OpenSER-Users] Openser 1.3 howto
by Fabio Isgro 12 Dec '07

12 Dec '07

How to install openser 1.3 there is any howto out there? Best Regards -- -\\_ Fabio Isgro' -}}\_ Babel S.r.l. - http://www.babel.it -}}}\-__ Tel. +39.06.91801075 }\/ -}}}\\°_\ Fax +39.06.91612446 }/\ }}}}\\ _/ Voip sip:9093@voip.babel.it ------ P.zza S.Benedetto da Norcia, 33 - 00040 Pomezia (Roma) B ° a ° b° e° l "CONFIDENZIALE: Questo messaggio ed i suoi allegati sono di carattere confidenziale per i destinatari in indirizzo. Se hai ricevuto questo messaggio per errore sei invitato cortesemente a rispondere immediatamente al mittente e cancellare tutti i suoi contenuti."

2 1

[OpenSER-Users] CPL DTD
by Daniel Grotti 12 Dec '07

12 Dec '07

Hi Bogdan, Cpl modules is based on DTD file. It doesn't support namespaces. So how could I make extensions (for example add a new node) on the dtd file remaining RFC3880 compliant ? Is it possible? Regards, daniel

1 0

[OpenSER-Users] Display Registered Users
by VoIP Forums www.Go4Calls.com 12 Dec '07

12 Dec '07

Hi All, Could you please guide us with right cmd to display the users which are registered successfully with openser? Regards, www.Go4Calls.Com VoIP Forums _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

2 2

[OpenSER-Users] Question about avp_aliases and script_flags
by Helmut Kuper 12 Dec '07

12 Dec '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, avpops doc say that avp spec for integer values is $avp(i[script_flags]:<ID>) Is it possible to set avp aliases to that values, e.g. avp_aliases="x=i1:10" avp_aliases="y=i2:10" ? When I do that, I got this error message in openser 1.3: Nov 14 16:00:42 [28355] ERROR:core:add_avp_galias: duplicate alias/avp entry Nov 14 16:00:42 [28355] ERROR:core:add_avp_galias_str: add global alias failed Nov 14 16:00:42 [28355] CRITICAL:core:yyerror: parse error (54,3-4): invalid AVP aliases regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHOxLU4tZeNddg3dwRArlWAKCd7mGq6K3yODf/8bvZRa9F/vcpugCfWsb7 Ale6eO7Xz8C0fnvyg5m4/n0= =tPWq -----END PGP SIGNATURE-----

2 9

[OpenSER-Users] Could someone help me with WeSIP ?
by Daniel Grotti 11 Dec '07

11 Dec '07

Hi all, I'm trying to use WeSIP but I don't be able to connect to "Manager" link. I receive a 500 Http error. This is my server.xml file: where I hide Proxy IP Address with PROXY_IP and in OpenserSER I have setted modparam("seas", "listen_sockets","localhost:5080") ***************************************************************************************** <Server port="8005" shutdown="SHUTDOWN" debug="0"> <Service name="Tomcat-Standalone"> <Connector className="org.apache.catalina.connector.http.HttpConnector" port="8080" minProcessors="5" maxProcessors="75" enableLookups="true" address="PROXY_IP" acceptCount="10" debug="0" /> <Connector className="com.voztele.sipservlet.connector.SipConnector" minProcessors="5" maxProcessors="75" enableLookups="true" acceptCount="10" debug="0" addresses="PROXY_IP:5060" pathName="com.voztele" protocol="udp" > <ExtraProperties> <Property key="com.voztele.javax.sip.SER_ADDRESS" value="localhost" /> <Property key="com.voztele.javax.sip.SER_PORT" value="5080" /> <Property key="com.voztele.STATISTICS" value="false" /> <Property key="javax.sip.STACK_NAME" value="app_server_one" /> <Property key="javax.sip.THREAD_POOL_SIZE" value="100" /> <Property key="com.voztele.javax.sip.THREAD_POOL_SIZE" value="100" /> <Property key="com.voztele.javax.sip.REENTRANT_LISTENER" value="true" /> <Property key="javax.sip.AUTOMATIC_DIALOG_SUPPORT" value="false" /> <Property key="com.voztele.javax.sip.TRACE_LEVEL" value="0" /> </ExtraProperties> </Connector> <Engine name="Standalone" defaultHost="PROXY_IP" debug="0" className="com.voztele.sipservlet.core.ConvergedEngine"> <Mapper className="org.apache.catalina.core.StandardEngineMapper" protocol="HTTP/1.1"/> <Mapper className="org.apache.catalina.core.StandardEngineMapper" protocol="HTTP/1.0"/> <Mapper className="com.voztele.sipservlet.core.EngineSipMapper" protocol="SIP/2.0"/> <Logger className="org.apache.catalina.logger.FileLogger" timestamp="true"/> <Realm className="org.apache.catalina.realm.MemoryRealm" /> <Host name="PROXY_IP" nameSip="PROXY_IP_5060" debug="0" appBase="wesipapps" port="5060" className="com.voztele.sipservlet.core.ConvergedHost" unpackWARs="true"> <Logger className="org.apache.catalina.logger.FileLogger" timestamp="true"/> <Mapper className="com.voztele.sipservlet.core.SipHostMapper" protocol="SIP/2.0"/> <Mapper className="org.apache.catalina.core.HttpHostMapper" protocol="HTTP/1.1"/> <Mapper className="org.apache.catalina.core.HttpHostMapper" protocol="HTTP/1.0"/>  </Host> </Engine> </Service> </Server> ************************************************************************************************************* Regards, daniel

2 1

[OpenSER-Users] Softphones supporting "pua_bla"?
by Iñaki Baz Castillo 11 Dec '07

11 Dec '07

Hi, I'd like to try the "pua_bla" module but don't have a SNOM or Polycom (models I think they support BLA). I've a Thomson S2030 but it does BLF (it doesn't send a REGISTER, just a SUBSCRIBE with "Event: dialog"). Is there any way to try the "pua_bla" module? any softphone supporting it? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

4 6

[OpenSER-Users] FW: No user and password are stored in the openxcap db, how do i do?
by KevinKinnan 11 Dec '07

11 Dec '07

1st question: There is no user and pwd in the db for the first time use. if I want to send a http request to openxcap what user name and password should i use? BTW, openims core and openser(presence module) are using in front of openxcap. I atteched a trace that still showed 401 after digesting request(by default, the username and password are both alice). 2nd question: I am designing a j2me programe to connect to openxcap, but only GET, POST, HEAD methods are supoorted in j2me. Can I use POST instead of PUT for request sending? THANK U! Regards, Kevin 使用新一代 Windows Live Messenger 轻松交流和共享！立即体验！ _________________________________________________________________ 天凉了，添衣了，心动了，“七件”了 http://get.live.cn

1 0

Re: [Serusers] Loading attributes question
by Jiri Kuthan 11 Dec '07

11 Dec '07

I'm afraid there is no easy way for doing so -- the multiple-value AVPs have been seen as "sets", disregarding the order. I think one would have to have a new function like order_avp($avp)... At 00:05 10/12/2007, Steven C. Blair wrote: >Content-Language: en-US >Content-Type: multipart/alternative; > boundary="_000_EBC0E89A333CED46A6273F415B6386E814BA4E6EEBMAILBOXexchan_" > > > >I have a locally defined table with attributes which may contain several values. I define each attribute /value pair as a new entry in the table then use avp_load to retrieve these values from the database. The problem is that I don't know how to retrieve the values in a specific order. For example if I have a=1, a=2, a=3 when I load the values for "a" they may be 2,1,3 or 3,1,2, etc. Is there any way to force loading in a specific sequence. > >Thanks,Steve >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

1 0

[OpenSER-Users] users@lists.openser.org
by minz1 11 Dec '07

11 Dec '07

Hi Schumann Sebastian, I too have the same problem regarding XCap Server project. ---------------------------------------------------------------------------------------- Following are the trace generated after running "python2.4 openxcap --no-fork" command:- ---------------------------------------------------------------------------------------- Starting Open XCAP 0.9.7 Traceback (most recent call last): File "/usr/local/bin/openxcap", line 56, in ? from xcap.server import XCAPServer File "/usr/local/lib/python2.4/site-packages/xcap/server.py", line 21, in ? from xcap import authentication File "/usr/local/lib/python2.4/site-packages/xcap/authentication.py", line 21, in ? from xcap.appusage import getApplicationForURI File "/usr/local/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 491, in ? applications = {'xcap-caps': XCAPCapabilitiesApplication(), File "/usr/local/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 64, in __init__ self.xml_schema = etree.XMLSchema(xml_schema_doc) File "xmlschema.pxi", line 67, in etree.XMLSchema.__init__ etree.XMLSchemaParseError: Document is not valid XML Schema ------------------------------------------------------------------------------------------ Following are the files which we are using:= ================================= openxcap-0.9.7 libxml2-2.6.30 twisted-web2-0.2.0+svn20070403.orig Python-2.4.2 lxml-1.3.4 zope.interface-3.3.0 python-application-1.0.9 MySQL-python-1.2.2 ================================= I followed the instructions from the openxcap webpage and did what instructed there but not able to make it work. Please help me. I will be always greatful to that help. regds Suraj Schumann Sebastian wrote: Dear all I have a big problem with openxcap. I don't get it work :( As I saw on their webpage, this mailing list can be posted for this concern as well. I am using version 0.9.3 and acc. the installation file on http://www.openxcap.org all current dependencies from the Debian unstable distribution. When I start /usr/bin/openxcap, the following error occurs in the syslog: Sep 25 17:01:21 openxcap openxcap[11876]: [-] Log opened. Sep 25 17:01:21 openxcap openxcap[11876]: [-] Starting Open XCAP 0.9.3 Sep 25 17:01:27 openxcap openxcap[11876]: [-] Traceback (most recent call last): Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/bin/openxcap", line 56, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap.server import XCAPServer Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/server.py", line 21, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap import authentication Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/authentication.py", line 21, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap.appusage import getApplicationForURI Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 466, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] applications = {'xcap-caps': XCAPCapabilitiesApplication(), Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 64, in __init__ Sep 25 17:01:27 openxcap openxcap[11876]: [-] self.xml_schema = etree.XMLSchema(xml_schema_doc) Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "xmlschema.pxi", line 67, in etree.XMLSchema.__init__ Sep 25 17:01:27 openxcap openxcap[11876]: [-] etree.XMLSchemaParseError: Document is not valid XML Schema As I wrote, I followed all instructions from the openxcap webpage but didn't find any hints how to make it work. Any help on that would be appreciated. Best regards Sebastian -- Sebastian Schumann Diploma Student Architecture and Design Department Slovak Telekom, a. s. T-Com, Operations Unit Production and Service Division Innovations and Enterprise Solution Subunit Address: Namestie Slobody 6, 817 62 Bratislava, Slovakia Office: Room 449, Jarabinkova 1, 821 09 Bratislava, Slovakia +421 2 588 13332 (tel) +421 910 643010 (mobile) +49 175 1925928 (mobile) -- View this message in context: http://www.nabble.com/users%40lists.openser.org-tp14279161p14279161.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

1 0

[OpenSER-Users] users@lists.openser.org
by minz1 11 Dec '07

11 Dec '07

Hi Schumann Sebastian, I too have the same problem regarding XCap Server project. ---------------------------------------------------------------------------------------- Following are the trace generated after running "python2.4 openxcap --no-fork" command:- ---------------------------------------------------------------------------------------- Starting Open XCAP 0.9.7 Traceback (most recent call last): File "/usr/local/bin/openxcap", line 56, in ? from xcap.server import XCAPServer File "/usr/local/lib/python2.4/site-packages/xcap/server.py", line 21, in ? from xcap import authentication File "/usr/local/lib/python2.4/site-packages/xcap/authentication.py", line 21, in ? from xcap.appusage import getApplicationForURI File "/usr/local/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 491, in ? applications = {'xcap-caps': XCAPCapabilitiesApplication(), File "/usr/local/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 64, in __init__ self.xml_schema = etree.XMLSchema(xml_schema_doc) File "xmlschema.pxi", line 67, in etree.XMLSchema.__init__ etree.XMLSchemaParseError: Document is not valid XML Schema ------------------------------------------------------------------------------------------ Following are the files which we are using:= ================================= openxcap-0.9.7 libxml2-2.6.30 twisted-web2-0.2.0+svn20070403.orig Python-2.4.2 lxml-1.3.4 zope.interface-3.3.0 python-application-1.0.9 MySQL-python-1.2.2 ================================= I followed the instructions from the openxcap webpage and did what instructed there but not able to make it work. Please help me. I will be always greatful to that help. regds Suraj Schumann Sebastian wrote: Dear all I have a big problem with openxcap. I don't get it work :( As I saw on their webpage, this mailing list can be posted for this concern as well. I am using version 0.9.3 and acc. the installation file on http://www.openxcap.org all current dependencies from the Debian unstable distribution. When I start /usr/bin/openxcap, the following error occurs in the syslog: Sep 25 17:01:21 openxcap openxcap[11876]: [-] Log opened. Sep 25 17:01:21 openxcap openxcap[11876]: [-] Starting Open XCAP 0.9.3 Sep 25 17:01:27 openxcap openxcap[11876]: [-] Traceback (most recent call last): Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/bin/openxcap", line 56, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap.server import XCAPServer Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/server.py", line 21, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap import authentication Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/authentication.py", line 21, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap.appusage import getApplicationForURI Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 466, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] applications = {'xcap-caps': XCAPCapabilitiesApplication(), Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 64, in __init__ Sep 25 17:01:27 openxcap openxcap[11876]: [-] self.xml_schema = etree.XMLSchema(xml_schema_doc) Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "xmlschema.pxi", line 67, in etree.XMLSchema.__init__ Sep 25 17:01:27 openxcap openxcap[11876]: [-] etree.XMLSchemaParseError: Document is not valid XML Schema As I wrote, I followed all instructions from the openxcap webpage but didn't find any hints how to make it work. Any help on that would be appreciated. Best regards Sebastian -- Sebastian Schumann Diploma Student Architecture and Design Department Slovak Telekom, a. s. T-Com, Operations Unit Production and Service Division Innovations and Enterprise Solution Subunit Address: Namestie Slobody 6, 817 62 Bratislava, Slovakia Office: Room 449, Jarabinkova 1, 821 09 Bratislava, Slovakia +421 2 588 13332 (tel) +421 910 643010 (mobile) +49 175 1925928 (mobile) -- View this message in context: http://www.nabble.com/users%40lists.openser.org-tp14279160p14279160.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

1 0

[Serusers] rewritehostport and RFC3261 compliant
by inge 11 Dec '07

11 Dec '07

Hi all, I find an issue with a problem encountered on some products. The problem only concerns on-net calls. In fact, when a SER User A calls a SER User B : - if B doesn't answer, after the tm timer : ==> t_on_failure("1") ==> setflagt(24) ==> load $ruri/username in vm_uri/usr_vm if flag is set to 24 then ==> avp_pushto $ruri s:vm_uri ==> resetflag(24) ==> add prefix Prefix("710") ==> rewritehostport(Asterisk:4060) ==> append_branch ==> t_relay_to_udp(Asterisk,4060) This is the process apply for transfer the call to the voicemail Asterisk. But, and only in this on-net case, the To-Tag change during this process, cause the failure of communication for some products. Whereas the RFC 3261 specify "... This tag will be incorporated by both endpoints into the dialog and will be included in all future requests and responses in this call ...". Thanks for your support.

3 6

[OpenSER-Users] new stuff in 1.3.0
by Daniel-Constantin Mierla 11 Dec '07

11 Dec '07

Hello, I started a dokuwiki page to collect the new things in openser 1.3.x, I still have a lot to add, however, since there are new modules and enhancements from many developers, it is better that everybody contribute to this page, to have a good image of what is new in the upcoming release. I am sure I will be forgetting some. The page will be used in building release announcement. http://www.openser.org/dokuwiki/doku.php/sandbox:new-in-1.3.x Thanks, Daniel

1 0

[Serusers] media proxy manage NAT IP Change?
by Gonzalo J. Sambucaro 11 Dec '07

11 Dec '07

Hi, I would like to know if the Media Proxy Server is able to manage the NAT IP change of an established rtp stream and continue working with the new NAT IP without problems? Saludos, Gonzalo. -- Gonzalo J. Sambucaro Ingeniería de Software Tel: +54-341-4808585 MSLC gonzalo(a)mslc.com.ar www.mslc.com.ar Ocampo y Esmeralda - Vivero de Empresas de Base Tecnológica Ciudad Universitaria Rosario UNR, CCT CONICET Rosario - Santa Fé - Argentina

1 0

[OpenSER-Users] trunk - openserro password not working
by Richard Bennett 11 Dec '07

11 Dec '07

Hi, After moving a working openser config for trunk to a new server, and updating the sources, I kept getting the error: ERROR:mysql:db_mysql_new_connection: driver error: Access denied for user 'openserro'@'localhost' (using password: YES) I double-checked, and the settings in: /usr/local/etc/openser/openserctlrc were set correct: DBROPW=newpassword So I set the mysql password for openserro back to the default 'openserro' in mysql, and it works, even with a different value in openserctlrc. Is this a known issue, bug, or am I doing something wrong? Richard.

2 3

[OpenSER-Users] Set CallerID as Username
by VoIP Forums www.Go4Calls.com 11 Dec '07

11 Dec '07

Hi All, i was routing calls from openser to asterisk to make accounting in asterisk side. I need small help, how i can set callerid = registrar username so in asterisk server i can capture the callerid to do the billing etc.... Thank You _________________________________________________________________ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+world&mkt=en-US&form=QBRE

2 1

Re: [OpenSER-Users] auth_db module - get_ha1(): Error while querying database
by Daniel-Constantin Mierla 11 Dec '07

11 Dec '07

Hello, seems that you haven't updated the database structure. In 1.2, there was a refurbishment that included primary keys, a.s.o. As I could guess from patch, you still use the old database schema. Cheers, Daniel PS. Please CC to mailing list, the answer can come much faster, with more competence from people experiencing same issues, and will help new comers with same case. On 12/10/07 15:07, Mihail Gluhowchenko wrote: > Hello; > Sorry %) > -- > Regards, > Mihail Gluhowchenko. > Enterprise ELTEX > 2007/12/10, Daniel-Constantin Mierla <daniel(a)voice-system.ro > <mailto:daniel@voice-system.ro>>: > > Hello, > > can you please send unified format for the patch? > > diff -u oldfile newfile > > Thanks, > Daniel > > > On 12/10/07 14:29, Mihail Gluhowchenko wrote: > > Hello; > > Here it is sc.patch; > > my version is 'openser-1.2.2-notls' > > -- > > Regards, > > Mihail Gluhowchenko. > > Enterprise ELTEX. > > > > 2007/12/10, Daniel-Constantin Mierla < daniel(a)voice-system.ro > <mailto:daniel@voice-system.ro> > > <mailto:daniel@voice-system.ro <mailto:daniel@voice-system.ro>>>: > > > > > > > > On 12/10/07 07:04, Mihail Gluhowchenko wrote: > > > Hello all work; :) > > was your version old or there is a bug in sc.dbtext? Can you > post the > > patch of changes? > > > > Best regards, > > Daniel > > > > > > > in script sc.dbtext use old format subscriber. > > > Little changes and all work > > > > > > date_now=$(date +"%s") > > > GLOB_ID=$(od -N4 -d /dev/random | head -n1 | cut -d' ' -f2,3 | > > sed -e > > > 's/ //') > > > > > > add) > > > if [ $# -ne 4 ] ; then > > > usage > > > exit 1 > > > fi > > > shift > > > credentials $1 $2 > > > is_user $1 > > > if [ $? -eq 0 ] ; then > > > echo -e "\nuser already exists\n" > > > exit 1 > > > fi > > > #Add to SUBSCRIBER file > > > #id(int,auto) username(str) domain(str) > > password(str) > > > first_name(str,null) last_name(str,null) email_address(str) > > > datetime_created(int) last_modified(int) ha1(str) ha1b(str) > > > rpid(str,null) > > > > > > > > > line="$GLOB_ID:$SERUSER:$SERDOMAIN:$2:::$3:$date_now:$date_now:$HA1:$HA1B:" > > > > > > echo $line >> $SUB_FILE > > > #Add to URI file > > > #id(int,auto) domain(str) uri_user(str) > > > last_modified(int) > > > line="$GLOB_ID:$SERDOMAIN:$SERUSER:$date_now" > > > echo $line >> $URI_FILE > > > echo -e "\nNew user ($SERUSER@$SERDOMAIN) > added to > > > DBTEXT files\n" > > > ;; > > > -- > > > Regards, > > > Mihail Gluhowchenko. > > > Enterprise ELTEX. > > > > > > > > >

2 1

[OpenSER-Users] Loop Detection
by Daniel Corbe 11 Dec '07

11 Dec '07

I have a problem. I'm taking INVITES from a PortaSIP proxy and handing it back with a new request URI and To: header; however, their loop detection algorithm is calling this a loop. Is there a way to fix this from within OpenSER? I'm assuming my proxy would have to send an INVITE with a completely new branch tag and From tag? 10 Dec 14:52:30.831/GLOBAL/b2bua: SENDING message to 206.71.169.114:5060: INVITE sip:*94310@206.71.169.114:5060 SIP/2.0 Via: SIP/2.0/UDP 216.151.143.69:5061 ;branch=z9hG4bKecaee2770d3db1e3a26ddc741511b55b;rport Max-Forwards: 70 From: <sip:18589520696@216.151.143.69>;tag=3b383c39c2bc93d8106f4425074ef399 To: <sip:*94310@206.71.169.114> Call-ID: 722814-3406315950-714741(a)msx69.mydomain.com~1o CSeq: 200 INVITE Contact: Anonymous <sip:216.151.143.69:5061> Expires: 300 User-Agent: Sippy cisco-GUID: 2108088383-2809270748-2290810900-572575638 h323-conf-id: 2108088383-2809270748-2290810900-572575638 Content-Length: 201 Content-Type: application/sdp v=0 o=Sippy 156402124 0 IN IP4 216.151.143.69 s=sip call t=0 0 m=audio 42486 RTP/AVP 18 0 8 c=IN IP4 216.151.143.77 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=sendrecv 10 Dec 14:52:30.841/GLOBAL/b2bua: RECEIVED message from 206.71.169.114:5060: SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 216.151.143.69:5061 ;branch=z9hG4bKecaee2770d3db1e3a26ddc741511b55b;rport=5061 From: <sip:18589520696@216.151.143.69>;tag=3b383c39c2bc93d8106f4425074ef399 To: <sip:*94310@206.71.169.114> Call-ID: 722814-3406315950-714741(a)msx69.mydomain.com~1o CSeq: 200 INVITE Server: OpenSER (1.2.2-notls (i386/freebsd)) Content-Length: 0 10 Dec 14:52:30.842/GLOBAL/ser[69804]: RECEIVED message from 206.71.169.114:5060: INVITE sip:1a10000000009@216.151.143.69:5060 SIP/2.0 Record-Route: <sip:206.71.169.114 ;lr=on;ftag=3b383c39c2bc93d8106f4425074ef399;did=fad.a2f4eb52> Via: SIP/2.0/UDP 206.71.169.114;branch=z9hG4bK961c.41ea57a.0 Via: SIP/2.0/UDP 216.151.143.69:5061 ;branch=z9hG4bKecaee2770d3db1e3a26ddc741511b55b;rport=5061 Max-Forwards: 69 From: <sip:18589520696@216.151.143.69>;tag=3b383c39c2bc93d8106f4425074ef399 To: <sip:1a10000000009@216.151.143.69:5060> Call-ID: 722814-3406315950-714741(a)msx69.mydomain.com~1o CSeq: 200 INVITE Contact: Anonymous <sip:216.151.143.69:5061> Expires: 300 User-Agent: Sippy cisco-GUID: 2108088383-2809270748-2290810900-572575638 h323-conf-id: 2108088383-2809270748-2290810900-572575638 Content-Length: 201 Content-Type: application/sdp P-hint: rr-enforced v=0 o=Sippy 156402124 0 IN IP4 216.151.143.69 s=sip call t=0 0 m=audio 42486 RTP/AVP 18 0 8 c=IN IP4 216.151.143.77 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=sendrecv 10 Dec 14:52:30.842/722814-3406315950-714741@msx69.mydomain.com~1o/ser[69804]: processing INVITE received from 206.71.169.114 10 Dec 14:52:30.843/GLOBAL/ser[69804]: SENDING message to 206.71.169.114:5060: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 206.71.169.114;branch=z9hG4bK961c.41ea57a.0 Via: SIP/2.0/UDP 216.151.143.69:5061 ;branch=z9hG4bKecaee2770d3db1e3a26ddc741511b55b;rport=5061 From: <sip:18589520696@216.151.143.69>;tag=3b383c39c2bc93d8106f4425074ef399 To: <sip:1a10000000009@216.151.143.69:5060> Call-ID: 722814-3406315950-714741(a)msx69.mydomain.com~1o CSeq: 200 INVITE Server: Sip EXpress router (0.9.6 (i386/freebsd)) Content-Length: 0 10 Dec 14:52:30.844/GLOBAL/ser[69804]: SENDING message to 216.151.143.69:5061: INVITE sip:1a10000000009@216.151.143.69:5061 SIP/2.0 Record-Route: <sip:216.151.143.69;ftag=3b383c39c2bc93d8106f4425074ef399;lr> Record-Route: <sip:206.71.169.114 ;lr=on;ftag=3b383c39c2bc93d8106f4425074ef399;did=fad.a2f4eb52> Via: SIP/2.0/UDP 216.151.143.69;branch= z9hG4bK961c.003d31266db19cfeb34aa550d186a976.0 Via: SIP/2.0/UDP 206.71.169.114;branch=z9hG4bK961c.41ea57a.0 Via: SIP/2.0/UDP 216.151.143.69:5061 ;branch=z9hG4bKecaee2770d3db1e3a26ddc741511b55b;rport=5061 Max-Forwards: 16 From: <sip:18589520696@216.151.143.69>;tag=3b383c39c2bc93d8106f4425074ef399 To: <sip:1a10000000009@216.151.143.69:5060> Call-ID: 722814-3406315950-714741(a)msx69.mydomain.com~1o CSeq: 200 INVITE Contact: Anonymous <sip:216.151.143.69:5061> Expires: 300 User-Agent: Sippy cisco-GUID: 2108088383-2809270748-2290810900-572575638 Content-Length: 201 Content-Type: application/sdp P-hint: rr-enforced v=0 o=Sippy 156402124 0 IN IP4 216.151.143.69 s=sip call t=0 0 m=audio 42486 RTP/AVP 18 0 8 c=IN IP4 216.151.143.77 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=sendrecv 10 Dec 14:52:30.844/GLOBAL/b2bua: RECEIVED message from 216.151.143.69:5060: INVITE sip:1a10000000009@216.151.143.69:5061 SIP/2.0 Record-Route: <sip:216.151.143.69;ftag=3b383c39c2bc93d8106f4425074ef399;lr> Record-Route: <sip:206.71.169.114 ;lr=on;ftag=3b383c39c2bc93d8106f4425074ef399;did=fad.a2f4eb52> Via: SIP/2.0/UDP 216.151.143.69;branch= z9hG4bK961c.003d31266db19cfeb34aa550d186a976.0 Via: SIP/2.0/UDP 206.71.169.114;branch=z9hG4bK961c.41ea57a.0 Via: SIP/2.0/UDP 216.151.143.69:5061 ;branch=z9hG4bKecaee2770d3db1e3a26ddc741511b55b;rport=5061 Max-Forwards: 16 From: <sip:18589520696@216.151.143.69>;tag=3b383c39c2bc93d8106f4425074ef399 To: <sip:1a10000000009@216.151.143.69:5060> Call-ID: 722814-3406315950-714741(a)msx69.mydomain.com~1o CSeq: 200 INVITE Contact: Anonymous <sip:216.151.143.69:5061> Expires: 300 User-Agent: Sippy cisco-GUID: 2108088383-2809270748-2290810900-572575638 Content-Length: 201 Content-Type: application/sdp P-hint: rr-enforced v=0 o=Sippy 156402124 0 IN IP4 216.151.143.69 s=sip call t=0 0 m=audio 42486 RTP/AVP 18 0 8 c=IN IP4 216.151.143.77 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=sendrecv 10 Dec 14:52:30.847/GLOBAL/b2bua: SENDING message to 216.151.143.69:5060: SIP/2.0 482 Loop Detected Via: SIP/2.0/UDP 216.151.143.69;branch= z9hG4bK961c.003d31266db19cfeb34aa550d186a976.0 Via: SIP/2.0/UDP 206.71.169.114;branch=z9hG4bK961c.41ea57a.0 Via: SIP/2.0/UDP 216.151.143.69:5061 ;branch=z9hG4bKecaee2770d3db1e3a26ddc741511b55b;rport=5061 Record-Route: <sip:216.151.143.69;ftag=3b383c39c2bc93d8106f4425074ef399;lr> Record-Route: <sip:206.71.169.114 ;lr=on;ftag=3b383c39c2bc93d8106f4425074ef399;did=fad.a2f4eb52> From: <sip:18589520696@216.151.143.69>;tag=3b383c39c2bc93d8106f4425074ef399 To: <sip:1a10000000009@216.151.143.69:5060> Call-ID: 722814-3406315950-714741(a)msx69.mydomain.com~1o CSeq: 200 INVITE Server: Sippy

2 5

Re: [OpenSER-Users] Problem with Mediaproxy
by Daniel 10 Dec '07

10 Dec '07

Hi, this is my call flow (from client side making the call: 192.168.10.224): The INVITE received by 192.168.0.167 (sent by 192.168.10.224 and processed by proxy) has SDP body with PRIVATE IP and PORT. ----------------------------------------------------------------------------------- # 192.162.10.224 ----> PROXY Request-Line: INVITE sip:franco.callegati@PROXY_IP SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.10.224:64244 ;branch=z9hG4bK-d87543-9947ff15e2520b54-1--d87543- Max-Forwards: 70 Contact: <sip:daniel.grotti@192.168.10.224:64244> To: "franco.callegati"<sip:franco.callegati@PROXY_IP> From: "casa"<sip:daniel.grotti@PROXY_IP>;tag=4554424c Call-ID: YjRmZjU3ODAwMjIxMDU0MDUwNDllYzUwNmZiYTJjMmE. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username="1000",realm="PROXY_IP",nonce="47593c9fa32034a66fdb7493ff499c7215d4d8db",uri="sip: franco.callegati@PROXY _IP",response="82d21af18029aa02388869088c3619d2",algorithm=MD5 User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 374 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 8 2 IN IP4 192.168.10.224 Session Name (s): CounterPath X-Lite 3.0 Connection Information (c): IN IP4 192.168.10.224 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 53624 RTP/AVP 107 119 100 106 0 105 98 8 101 Media Attribute (a): fmtp:101 0-15 Media Attribute (a): rtpmap:107 BV32/16000 Media Attribute (a): rtpmap:119 BV32-FEC/16000 Media Attribute (a): rtpmap:100 SPEEX/16000 Media Attribute (a): rtpmap:106 SPEEX-FEC/16000 Media Attribute (a): rtpmap:105 SPEEX-FEC/8000 Media Attribute (a): rtpmap:98 iLBC/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): sendrecv # PROXY -----> 192.162.10.224 Status-Line: SIP/2.0 100 Trying Message Header Via: SIP/2.0/UDP 192.168.10.224:64244 ;branch=z9hG4bK-d87543-9947ff15e2520b54-1--d87543-;rport=28943;received= 85.18.14.25 To: "franco.callegati"<sip:franco.callegati@PROXY_IP> From: "casa"<sip:daniel.grotti@PROXY_IP>;tag=4554424c Call-ID: YjRmZjU3ODAwMjIxMDU0MDUwNDllYzUwNmZiYTJjMmE. CSeq: 2 INVITE Server: OpenSER (1.3.0-pre1-notls (i386/linux)) Content-Length: 0 # PROXY -----> 192.162.10.224 Status-Line: SIP/2.0 180 Ringing Message Header Via: SIP/2.0/UDP 192.168.10.224:64244;received=85.18.14.25 ;branch=z9hG4bK-d87543-9947ff15e2520b54-1--d87543-;rport=28943 Record-Route: <sip:PROXY_IP:5060;lr;nat=yes;ftag=4554424c> Contact: <sip:franco.callegati@85.18.14.25 :42214;rinstance=96888d44783df4a0> To: "franco.callegati"<sip:franco.callegati@PROXY_IP>;tag=d06d8b2d From: "casa"<sip:daniel.grotti@PROXY_IP>;tag=4554424c Call-ID: YjRmZjU3ODAwMjIxMDU0MDUwNDllYzUwNmZiYTJjMmE. CSeq: 2 INVITE User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 # PROXY -----> 192.162.10.224 Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.10.224:64244;received=85.18.14.25 ;branch=z9hG4bK-d87543-9947ff15e2520b54-1--d87543-;rport=28943 Record-Route: <sip:PROXY_IP:5060;lr;nat=yes;ftag=4554424c> Contact: <sip:franco.callegati@85.18.14.25 :42214;rinstance=96888d44783df4a0> To: "franco.callegati"<sip:franco.callegati@PROXY_IP>;tag=d06d8b2d From: "casa"<sip:daniel.grotti@PROXY_IP>;tag=4554424c Call-ID: YjRmZjU3ODAwMjIxMDU0MDUwNDllYzUwNmZiYTJjMmE. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 372 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 7 2 IN IP4 192.168.0.167 Session Name (s): CounterPath X-Lite 3.0 Connection Information (c): IN IP4 192.168.0.167 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 45566 RTP/AVP 107 119 100 106 0 105 98 8 101 Media Attribute (a): fmtp:101 0-15 Media Attribute (a): rtpmap:107 BV32/16000 Media Attribute (a): rtpmap:119 BV32-FEC/16000 Media Attribute (a): rtpmap:100 SPEEX/16000 Media Attribute (a): rtpmap:106 SPEEX-FEC/16000 Media Attribute (a): rtpmap:105 SPEEX-FEC/8000 Media Attribute (a): rtpmap:98 iLBC/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): sendrecv 192.162.10.224 ----> PROXY Request-Line: ACK sip:franco.callegati@85.18.14.25:42214;rinstance=96888d44783df4a0 SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.10.224:64244 ;branch=z9hG4bK-d87543-66641951d8357408-1--d87543- Max-Forwards: 70 Route: <sip:PROXY_IP:5060;lr;nat=yes;ftag=4554424c> Contact: <sip:daniel.grotti@192.168.10.224:64244> To: "franco.callegati"<sip:franco.callegati@PROXY_IP>;tag=d06d8b2d From: "casa"<sip:daniel.grotti@PROXY_IP>;tag=4554424c Call-ID: YjRmZjU3ODAwMjIxMDU0MDUwNDllYzUwNmZiYTJjMmE. CSeq: 2 ACK Proxy-Authorization: Digest username="1000",realm="PROXY_IP",nonce="47593c9fa32034a66fdb7493ff499c7215d4d8db",uri="sip: franco.callegati@PROXY _IP",response="82d21af18029aa02388869088c3619d2",algorithm=MD5 User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 RTP STREAM BETWEEN CLIENTS (MEDIAPROXY DOESN'T WORK) CALL ENDED ---------------------------------------------------------------------------------------------------------------------- this is my config file: debug=3 fork=yes log_stderror=yes listen=PROXY_IP port=5060 children=4 dns=no rev_dns=no disable_dns_blacklist=yes mpath="/usr/SER_proxy//lib/openser/modules/" #------------------------------------------------------ # Load Module #------------------------------------------------------ loadmodule "mysql.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "mi_fifo.so" loadmodule "uri.so" loadmodule "uri_db.so" loadmodule "domain.so" loadmodule "mediaproxy.so" loadmodule "nathelper.so" loadmodule "textops.so" loadmodule "xlog.so" loadmodule "cpl-c.so" loadmodule "alias_db.so" #---------------------------------------------------------- # Parameters #---------------------------------------------------------- modparam("auth_db|uri_db|usrloc|domain|alias_db", "db_url", " mysql://topenser:openserrw@localhost/topenser") modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("usrloc", "nat_bflag", 6) modparam("rr", "enable_full_lr", 1) modparam("alias_db", "use_domain", 1) # REGISTRAR MODULE modparam("registrar", "max_contacts", 2) # max number of contacts per URI modparam("registrar", "append_branches", 0) # if 0: R-URI will be overwritten with the highest-q rated contact and the rest will be left unprocessed. modparam("registrar", "default_q", 1000) # default contacts q-value set as 1 modparam("registrar", "received_avp","$avp(i:42)") # URI_DB MODULE modparam("uri_db", "use_uri_table", 1) # Specify if the "uri" table should be used for checkings instead of "subscriber" table # CPL MODULE modparam("cpl-c", "db_url", "mysql://topenser:openserrw@localhost/topenser") modparam("cpl-c", "cpl_table", "cpl") modparam("cpl-c", "cpl_dtd_file", "/usr/src/openser-1.2.2-notls /modules/cpl-c/cpl-06.dtd") modparam("cpl-c", "lookup_domain", "location") modparam("cpl-c","lookup_append_branches",1) modparam("cpl-c","proxy_route", 10) #NATHELPER MODULE modparam("nathelper", "rtpproxy_disable_tout", 120) modparam("nathelper", "rtpproxy_tout", 2) modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "received_avp", "$avp(i:42)") modparam("nathelper", "sipping_method", "OPTIONS") # Use OPTIONS every 10sec to keep NAT alive modparam("nathelper", "sipping_bflag", 6) modparam("nathelper", "sipping_from", "sip:PROXY_IP") #MEDIAPROXY MODULE modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy", "sip_asymmetrics", "/usr/local/etc/openser/sip-asymmetric-clients") modparam("mediaproxy", "rtp_asymmetrics", "/usr/local/etc/openser/rtp-asymmetric-clients") # TM MODULE modparam("tm", "fr_inv_timer", 120) #------------------------------------------------------------ #MAIN ROUTE #----------------------------------------------------------- route { xlog("L_DBG", "INFO: NEW REQUEST - METHOD=$rm R-URI=$ru FROM=$fu TO=$tu IP=$si ID=$ci\n"); #------------------------------------ # Sanity checks #------------------------------------ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too much Hops"); exit; }; if (msg:len >= 2048) { sl_send_reply("513", "Message too big"); exit; }; #------------------------------------ # RECORD-ROUTE #------------------------------------ if(method=="INVITE" && client_nat_test("7")) { record_route_preset ("PROXY_IP:5060;nat=yes"); xlog("L_DBG", "INFO: RECORD-ROUTE WITH NAT"); } else if (method!="REGISTER") { record_route(); }; #---------------------------------------- # CALL TEAR DOWN #---------------------------------------- if (method=="BYE") { end_media_session(); }; if (method=="CANCEL") { end_media_session(); }; #-------------------------------------------- # LOOSE ROUTE #-------------------------------------------- if (loose_route()) { xlog("L_DBG", "INFO: LOOSE ROUTE"); if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403","Unauthorized"); exit; }; consume_credentials(); if (client_nat_test("7") || search("^Route:.*nat=yes")) { setbflag(7); use_media_proxy(); }; }; route(1); exit; }; #------------------------------------------------- #CALL TYPE PROCESSING #------------------------------------------------- if (!uri==myself) { route(1); exit; }; if (uri==myself) { if (method=="REGISTER") { xlog("L_DBG", "INFO: REGISTER Message received"); route(2); exit; }; if (method=="ACK") { route(1); exit; }; if (method=="INVITE") { xlog("L_DBG", "INFO: INVITE Message received"); route(3); exit; }; if (method=="MESSAGE") { xlog("L_DBG", "INFO: MESSAGE method received"); route(4); exit; }; lookup("aliases"); if (!uri==myself) { route(1); exit; }; if (!lookup("location")) { sl_send_reply("404", "Destination not found"); exit; }; }; route(1); } #-------------------------------------------------------- # REGISTER PROCESSING #-------------------------------------------------------- route[2] { # Check credentials if(!www_authorize("","subscriber")) { www_challenge("","0"); exit; } else if(!check_to()) { sl_send_reply("401", "Unauthorized"); exit; }; consume_credentials(); # Check client natted - if yes, set flag 6 if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setbflag(6); xlog("L_DBG","INFO: Registration behind NAT"); xlog("L_DBG","INFO: Bflag 6, related to BRANCH=$T_branch_idx, is setted as $bf"); fix_nated_register(); force_rport(); }; if(!save("location")) { sl_reply_error(); xlog("L_DBG", "Registration of [$fU] Failed"); }; xlog("L_DBG", "INFO: Registration of [$fU] Complete."); xlog("L_DBG", "INFO: 200 OK has been sended"); } #---------------------------------------------------------- # INVITE PROCESSING #---------------------------------------------------------- route[3] { sl_send_reply("100", "Trying"); # Check if caller is natted, if yes set flag 7, rewrite Contact HF with Proxy address if(client_nat_test("7")) { setbflag(7); xlog("L_DBG", "INFO: CLIENT INVITE BEHIND NAT"); xlog("L_DBG","NOTICE: Bflag 7, related to BRANCH=$T_branch_idx of this INVITE, is setted as $bf"); if(force_rport()) { xlog("L_DBG","INFO: rport FORCED"); }; if(fix_nated_contact()) { xlog("L_DBG","INFO: Contact HF is now [$ct]"); }; # append_hf("P-hint: INVITE behind NAT \n"); }; # Check Credentials if(!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Unauthorized"); exit; }; lookup("aliases"); if(uri!=myself) { route(1); exit; }; consume_credentials(); # Check if r-URI is an alias. If yes, rewrite RURI with User's URI if(!alias_db_lookup("dbaliases")) { xlog("L_DBG", "The [$ru] is not an alias"); }; # Use Caller CPL "Outgoing" Script, if exist if(!cpl_run_script("outgoing", "force_stateful")) { sl_send_reply("500","Outgoing CPL script execution failed"); exit; }; # Find destination user and re-write R-URI with its Contact if (!lookup("location")) { sl_send_reply("404", "Destination not found"); exit; }; xlog("L_DBG", "INFO: LOCATION HAS BEEN FOUND"); xlog("L_DBG","INFO: R-URI is now [$ru]"); t_on_reply("1"); # Use CPL "Incoming" script of the destination client if(!cpl_run_script("incoming","force_stateful")) { sl_send_reply("500", "Incoming CPL script execution failed"); exit; }; route(11); route(1); } # -------------------------------- # PROXY ROUTE - CPL SCRIPT LOADED # -------------------------------- route[10] { xlog("L_DBG", "INFO: ROUTE 10 - PROXY ROUTE \n"); # append_hf("P-hint: CPL ROUTE 10 - PROXY-ROUTE\n"); if (isbflagset(6) || isbflagset(7)) { xlog("L_DBG", "INFO: flag 6 or 7 setted \n"); if(!isbflagset(8)) { setbflag(8); }; if(use_media_proxy()) { xlog("L_DBG","NOTICE: MEDIAPROXY in use for the CALLER!!"); # append_hf("P-hint: Mediaproxy in USE for the Caller\n"); }; }; } # ----------------------------- # CONTROL FLAG - NO CPL SCRIPT # ----------------------------- route[11] { xlog("L_DBG", "INFO: ROUTE 11 - NO CPL SCRIPT"); # append_hf("P-hint: ROUTE 11 \n"); if(isbflagset(6) || isbflagset(7)) { xlog("L_DBG", "INFO: flag 6 or 7 setted"); if(!isbflagset(8)) { setbflag(8); }; if(use_media_proxy()) { xlog("L_DBG","NOTICE: mediaproxy in use for the caller!!"); # append_hf("P-hint: MEDIAPROXY in USE for the CALLER\n"); }; }; } # -------------------------------------------------- # MESSAGE PROCESSING # -------------------------------------------------- route[4] { if (uri!=myself) { sl_send_reply ("403", "Forbidden"); exit; }; if (!lookup("location")) { sl_send_reply("404", "Destination not found"); exit; }; route(1); } #---------------------------------------------------- # RELEY ROUTE #---------------------------------------------------- route[1] { if (!t_relay("0x01")) # don't send 100 trying { if (method=="INVITE" || method=="ACK") { end_media_session(); xlog("L_DBG","INFO: MEDIA SESSION ENDED"); }; sl_reply_error(); xlog("L_DBG", "INFO: SENT ERROR REPLY"); }; } #--------------------------------------------------------------------- # On Reply Route #--------------------------------------------------------------------- onreply_route[1] { xlog("L_DBG","INFO : ONREPLY ROUTE"); # append_hf("P-hint: ONREPLY ROUTE \n"); if (status=~"(180)|(183)|2[0-9][0-9]") { if (isbflagset(8) && !search("^Content-Lenght:[ ]*0")) { if(use_media_proxy()) { xlog("L_DBG","INFO: MEDIAPROXY in use for the CALLED!"); # append_hf("P-hint: Media Proxy in use for the Callee!\n"); }; }; }; if (client_nat_test("7") && status=~"(180)|(183)|(200)") { fix_nated_contact(); xlog("L_DBG","INFO: CONTACT HF has been CHANGED"); # append_hf("P-hint: CALLEE'S CONTACT HF has benn CHANGED\n"); }; } ---------------------------------------------------------------------------------------------------------------------- Regards, Daniel Hi Daniel, Can u post your openser configuration as well as a complete ngrep trace for the calls you have problems with? DanB -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

3 3

[Serusers] ser and rtpproxy bridging
by mashudi 10 Dec '07

10 Dec '07

Hi all, I'm going to setup ser 0.9.6 for bridging two vpn network 10.12.12.0/24 and 10.177.10.0/24 using rtpproxy and a pstn gateway 10.177.10.60 , if some one can have any sample configuration of some good guide to configure. Please help. best regards, mashudi

1 0

[OpenSER-Users] openxcap Problem
by Schumann Sebastian 10 Dec '07

10 Dec '07

Dear all I have a big problem with openxcap. I don't get it work :( As I saw on their webpage, this mailing list can be posted for this concern as well. I am using version 0.9.3 and acc. the installation file on http://www.openxcap.org all current dependencies from the Debian unstable distribution. When I start /usr/bin/openxcap, the following error occurs in the syslog: Sep 25 17:01:21 openxcap openxcap[11876]: [-] Log opened. Sep 25 17:01:21 openxcap openxcap[11876]: [-] Starting Open XCAP 0.9.3 Sep 25 17:01:27 openxcap openxcap[11876]: [-] Traceback (most recent call last): Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/bin/openxcap", line 56, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap.server import XCAPServer Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/server.py", line 21, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap import authentication Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/authentication.py", line 21, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap.appusage import getApplicationForURI Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 466, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] applications = {'xcap-caps': XCAPCapabilitiesApplication(), Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 64, in __init__ Sep 25 17:01:27 openxcap openxcap[11876]: [-] self.xml_schema = etree.XMLSchema(xml_schema_doc) Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "xmlschema.pxi", line 67, in etree.XMLSchema.__init__ Sep 25 17:01:27 openxcap openxcap[11876]: [-] etree.XMLSchemaParseError: Document is not valid XML Schema As I wrote, I followed all instructions from the openxcap webpage but didn't find any hints how to make it work. Any help on that would be appreciated. Best regards Sebastian -- Sebastian Schumann Diploma Student Architecture and Design Department Slovak Telekom, a. s. T-Com, Operations Unit Production and Service Division Innovations and Enterprise Solution Subunit Address: Namestie Slobody 6, 817 62 Bratislava, Slovakia Office: Room 449, Jarabinkova 1, 821 09 Bratislava, Slovakia +421 2 588 13332 (tel) +421 910 643010 (mobile) +49 175 1925928 (mobile) sebastian.schumann(a)t-com.sk http://www.t-com.sk <http://www.t-com.sk/>

4 3

Re: [OpenSER-Users] auth_db module - get_ha1(): Error while querying database
by Daniel-Constantin Mierla 10 Dec '07

10 Dec '07

Hello, can you please send unified format for the patch? diff -u oldfile newfile Thanks, Daniel On 12/10/07 14:29, Mihail Gluhowchenko wrote: > Hello; > Here it is sc.patch; > my version is 'openser-1.2.2-notls' > -- > Regards, > Mihail Gluhowchenko. > Enterprise ELTEX. > > 2007/12/10, Daniel-Constantin Mierla < daniel(a)voice-system.ro > <mailto:daniel@voice-system.ro>>: > > > > On 12/10/07 07:04, Mihail Gluhowchenko wrote: > > Hello all work; :) > was your version old or there is a bug in sc.dbtext? Can you post the > patch of changes? > > Best regards, > Daniel > > > > in script sc.dbtext use old format subscriber. > > Little changes and all work > > > > date_now=$(date +"%s") > > GLOB_ID=$(od -N4 -d /dev/random | head -n1 | cut -d' ' -f2,3 | > sed -e > > 's/ //') > > > > add) > > if [ $# -ne 4 ] ; then > > usage > > exit 1 > > fi > > shift > > credentials $1 $2 > > is_user $1 > > if [ $? -eq 0 ] ; then > > echo -e "\nuser already exists\n" > > exit 1 > > fi > > #Add to SUBSCRIBER file > > #id(int,auto) username(str) domain(str) > password(str) > > first_name(str,null) last_name(str,null) email_address(str) > > datetime_created(int) last_modified(int) ha1(str) ha1b(str) > > rpid(str,null) > > > > > line="$GLOB_ID:$SERUSER:$SERDOMAIN:$2:::$3:$date_now:$date_now:$HA1:$HA1B:" > > > echo $line >> $SUB_FILE > > #Add to URI file > > #id(int,auto) domain(str) uri_user(str) > > last_modified(int) > > line="$GLOB_ID:$SERDOMAIN:$SERUSER:$date_now" > > echo $line >> $URI_FILE > > echo -e "\nNew user ($SERUSER@$SERDOMAIN) added to > > DBTEXT files\n" > > ;; > > -- > > Regards, > > Mihail Gluhowchenko. > > Enterprise ELTEX. > > > >

1 0

[Serusers] Problem for compiling with LCR
by Gerson A. Matiolli 10 Dec '07

10 Dec '07

Hi all I am compiling the SER-0.96 that downloaded the cvs with following command: make include_modules="mysql lcr" modules The following error occurred: make[1]: Entrando no diretÃ³rio `/home/ser/sip_router/modules/mysql' make[1]: `mysql.so' estÃ¡ atualizado. make[1]: Saindo do diretÃ³rio `/home/ser/sip_router/modules/mysql' make[1]: Entrando no diretÃ³rio `/home/ser/sip_router/modules/lcr' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -minline-all-stringops -falign-loops -ftree-vectorize -mtune=athlon64 -Wall -DNAME='"ser"' -DVERSION='"2.1.0-dev13"' -DARCH='"i386"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.1.2"' -D__CPU_i386 -D__OS_linux -DSER_VER=2001000 -DCFG_DIR='"/usr/local/etc/ser/"' -DMODS_DIR='"/usr/local/lib/ser/modules/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DUSE_TLS -DTLS_HOOKS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT -c lcr_mod.c -o lcr_mod.o In file included from ../mysql/dbase.h:42, from lcr_mod.c:53: ../mysql/../../db/db_val.h:40: error: redeclaration of enumerator âDB_INTâ ../../db/db_fld.h:47: error: previous definition of âDB_INTâ was here ../mysql/../../db/db_val.h:41: error: redeclaration of enumerator âDB_FLOATâ ../../db/db_fld.h:48: error: previous definition of âDB_FLOATâ was here ../mysql/../../db/db_val.h:42: error: redeclaration of enumerator âDB_DOUBLEâ ../../db/db_fld.h:49: error: previous definition of âDB_DOUBLEâ was here ../mysql/../../db/db_val.h:44: error: redeclaration of enumerator âDB_STRâ ../../db/db_fld.h:51: error: previous definition of âDB_STRâ was here ../mysql/../../db/db_val.h:45: error: redeclaration of enumerator âDB_DATETIMEâ ../../db/db_fld.h:52: error: previous definition of âDB_DATETIMEâ was here ../mysql/../../db/db_val.h:46: error: redeclaration of enumerator âDB_BLOBâ ../../db/db_fld.h:53: error: previous definition of âDB_BLOBâ was here ../mysql/../../db/db_val.h:48: error: redeclaration of enumerator âDB_BITMAPâ ../../db/db_fld.h:55: error: previous definition of âDB_BITMAPâ was here lcr_mod.c:118: error: expected â=â, â,â, â;â, âasmâ or â__attribute__â before âlcr_dbfâ lcr_mod.c: In function âlcr_db_initâ: lcr_mod.c:242: error: âlcr_dbfâ undeclared (first use in this function) lcr_mod.c:242: error: (Each undeclared identifier is reported only once lcr_mod.c:242: error: for each function it appears in.) lcr_mod.c: In function âlcr_db_bindâ: lcr_mod.c:260: warning: implicit declaration of function âbind_dbmodâ lcr_mod.c:260: error: âlcr_dbfâ undeclared (first use in this function) lcr_mod.c:266: warning: implicit declaration of function âDB_CAPABILITYâ lcr_mod.c:266: error: âDB_CAP_QUERYâ undeclared (first use in this function) lcr_mod.c:272: error: âDB_CAP_RAW_QUERYâ undeclared (first use in this function) lcr_mod.c: In function âlcr_db_closeâ: lcr_mod.c:284: error: âlcr_dbfâ undeclared (first use in this function) lcr_mod.c: In function âlcr_db_verâ: lcr_mod.c:296: error: âlcr_dbfâ undeclared (first use in this function) lcr_mod.c:305: warning: implicit declaration of function âtable_versionâ lcr_mod.c: In function âreload_gwsâ: lcr_mod.c:503: error: âdb_row_tâ undeclared (first use in this function) lcr_mod.c:503: error: ârowâ undeclared (first use in this function) lcr_mod.c:518: error: âlcr_dbfâ undeclared (first use in this function) lcr_mod.c:534: warning: implicit declaration of function âRES_ROW_Nâ lcr_mod.c:542: warning: implicit declaration of function âRES_ROWSâ lcr_mod.c:543: warning: implicit declaration of function âROW_VALUESâ lcr_mod.c:543: error: invalid type argument of â->â lcr_mod.c:549: error: invalid type argument of â->â lcr_mod.c:550: error: invalid type argument of â->â lcr_mod.c:553: error: invalid type argument of â->â lcr_mod.c:561: error: invalid type argument of â->â lcr_mod.c:564: error: invalid type argument of â->â lcr_mod.c:572: error: invalid type argument of â->â lcr_mod.c:575: error: invalid type argument of â->â lcr_mod.c:584: error: invalid type argument of â->â lcr_mod.c:588: error: invalid type argument of â->â lcr_mod.c: In function âload_gwsâ: lcr_mod.c:636: error: âdb_row_tâ undeclared (first use in this function) lcr_mod.c:636: error: ârowâ undeclared (first use in this function) lcr_mod.c:636: error: ârâ undeclared (first use in this function) lcr_mod.c:636: warning: left-hand operand of comma expression has no effect lcr_mod.c:698: error: âlcr_dbfâ undeclared (first use in this function) lcr_mod.c:705: error: invalid type argument of â->â lcr_mod.c:709: error: invalid type argument of â->â lcr_mod.c:712: error: invalid type argument of â->â lcr_mod.c:714: error: invalid type argument of â->â lcr_mod.c:717: error: invalid type argument of â->â lcr_mod.c:719: error: invalid type argument of â->â lcr_mod.c:722: error: invalid type argument of â->â lcr_mod.c:724: error: invalid type argument of â->â lcr_mod.c:727: error: invalid type argument of â->â lcr_mod.c:729: error: invalid type argument of â->â lcr_mod.c:733: error: invalid type argument of â->â make[1]: ** [lcr_mod.o] Erro 1 make[1]: Saindo do diretÃ³rio `/home/ser/sip_router/modules/lcr' make: ** [modules] Erro 1 Somebody help me Tanks

1 0

[Serusers] Loading attributes question
by Steven C. Blair 10 Dec '07

10 Dec '07

I have a locally defined table with attributes which may contain several values. I define each attribute /value pair as a new entry in the table then use avp_load to retrieve these values from the database. The problem is that I don't know how to retrieve the values in a specific order. For example if I have a=1, a=2, a=3 when I load the values for "a" they may be 2,1,3 or 3,1,2, etc. Is there any way to force loading in a specific sequence. Thanks,Steve

1 0

Re: [OpenSER-Users] auth_db module - get_ha1(): Error while querying database
by Daniel-Constantin Mierla 10 Dec '07

10 Dec '07

On 12/10/07 07:04, Mihail Gluhowchenko wrote: > Hello all work; :) was your version old or there is a bug in sc.dbtext? Can you post the patch of changes? Best regards, Daniel > in script sc.dbtext use old format subscriber. > Little changes and all work > > date_now=$(date +"%s") > GLOB_ID=$(od -N4 -d /dev/random | head -n1 | cut -d' ' -f2,3 | sed -e > 's/ //') > > add) > if [ $# -ne 4 ] ; then > usage > exit 1 > fi > shift > credentials $1 $2 > is_user $1 > if [ $? -eq 0 ] ; then > echo -e "\nuser already exists\n" > exit 1 > fi > #Add to SUBSCRIBER file > #id(int,auto) username(str) domain(str) password(str) > first_name(str,null) last_name(str,null) email_address(str) > datetime_created(int) last_modified(int) ha1(str) ha1b(str) > rpid(str,null) > > line="$GLOB_ID:$SERUSER:$SERDOMAIN:$2:::$3:$date_now:$date_now:$HA1:$HA1B:" > echo $line >> $SUB_FILE > #Add to URI file > #id(int,auto) domain(str) uri_user(str) > last_modified(int) > line="$GLOB_ID:$SERDOMAIN:$SERUSER:$date_now" > echo $line >> $URI_FILE > echo -e "\nNew user ($SERUSER@$SERDOMAIN) added to > DBTEXT files\n" > ;; > -- > Regards, > Mihail Gluhowchenko. > Enterprise ELTEX. >

1 0

[OpenSER-Users] Server Crash - Sending 500
by Kelvin Williams 10 Dec '07

10 Dec '07

While simulating a database failure (stopping the service), I noticed that OpenSER would also crash (see log below). While I don't mind the OpenSER crash since in our configuration its highly dependent upon the database, what I would like to do is send a 5xx series message to the client so that it would be forced to look at one of our other proxies to complete its call. As the client continued to send the INVITE to the original server. How can I send a 500 Server Internal Error message back to the client in the event of a database failure? Dec 8 22:36:11 terra /usr/local/sbin/openser[8134]: PG[get_result]: 0x81af628 Warning: Probable invalid query Dec 8 22:36:11 terra /usr/local/sbin/openser[8134]: PG[get_result]: 0x81af628 Warning: PGRES_FATAL_ERROR Dec 8 22:36:11 terra /usr/local/sbin/openser[8134]: PG[get_result]: 0x81af628 Warning: server closed the connection unexpectedly This probably means the server terminated abnormally before or while processing the request. Dec 8 22:36:11 terra /usr/local/sbin/openser[8147]: ERROR: receive_fd: EOF on 23 Dec 8 22:36:11 terra /usr/local/sbin/openser[8105]: child process 8134 exited by a signal 11 Dec 8 22:36:11 terra /usr/local/sbin/openser[8105]: core was not generated Dec 8 22:36:11 terra /usr/local/sbin/openser[8105]: INFO: terminating due to SIGCHLD Dec 8 22:36:11 terra /usr/local/sbin/openser[8147]: INFO: signal 15 received Dec 8 22:36:11 terra /usr/local/sbin/openser[8140]: INFO: signal 15 received Dec 8 22:36:11 terra /usr/local/sbin/openser[8137]: INFO: signal 15 received

3 2

[OpenSER-Users] Timeout between 100 and 180/183 Messages
by Dioris Moreno 10 Dec '07

10 Dec '07

I know fr_timer controls the timeout for provisional responses, and fr_inv_timer controls the timeout for final responses. Also, the TM Module documentation explains: "fr_timer - this timer is used when no response was received yet. If there is no response after fr_timer seconds the timer triggers (and failure route will be executed if t_on_failure() was called). If a provisional response was received, the timer is set to fr_inv_timer for INVITE transactions, and RT_T2 for all other transactions. If a final response is received, the transaction has finished." I understand this "If a provisional response was received, the timer is set to fr_inv_timer for INVITE", but how can I set a different timeout for cases when I have received 100 Trying and I didn't get a 180/183 Ringing? This is the example mentioned in documentation: "For example: You want to have fail over if there is no provisional response after 3 seconds, but you want to ring for 60 seconds. Thus, set the fr_timer to 3 and fr_inv_timer to 60." How can I get a fail over if there is no provisional response after 3 seconds, a fail over if there is no 180/183 message after receiving 100 Trying of 10 seconds, and I want the SIP client rings for only 60 seconds? I'll appreciate any help Dioris Moreno

2 1

Re: [Serusers] SIP Express Bundle - let's get started
by Mike Trest - Personal 10 Dec '07

10 Dec '07

To All, I offer comments / suggestion from perspective of enterprise user - more precisely from perspective of the IT-folks who will be installing and operating on behalf of their own enterprise. First, no matter what we select, we will not be able to satisfy those whose policy (or learning limitations) restrict them to a specific OS / DISTRO / VERSION / VENDOR. Just as there are myriads of enterprise with unique approaches to their particular industry, so also most have a very specific opinion about what works best for them. Therefore, I suggest we boldly assert, that SER is to be promoted as an appliance where the interests of the harried IT-folks can be best served. Let this bundle be delivered as a as a stand alone CD/DVD that, once it is deployed on a "buy it anywhere" machine, the IT-folks will never need to go beyond the external web / administrator interface into the inner realms of OS dependencies. <DREAM> This may be idealistic! However, broad scale enterprise acceptance can be achieved very quickly when we are able to achieve this goal. Oh, by-the-way, this broad user base who love the basic appliance for it's simplicity will also love those whose talents can deliver upscale versions of the same fundamental appliance. As one with enough grey hair and grandchildren to have loved my 195x vintage black and white television, I absolutely love the HD flat-screens of 2007! </DREAM> IMHO we should focus on an Enterprise level DISTRO. I agree with the statement "... I think the ultimate goal should be something like an ISO that's ready to go on a server..." Furthermore, I think virtual machine distributions are just another in a long line of "Silver Bullets" that may appeal to the developer heart in all of us . . . but which will scare the crap out of the policy focused Enterprise IT-folks. Therefore, I feel there is enough talent among us to create the first bundle in such a way to make it available to multiple Enterprise users so long as they understand it as an appliance! I have stated my preference for the CentOS-RHEL-FCx environment because it has the broadest base of enterprise-quality custom applications which equire mission-critical long-term stability. On the other hand, I will be supportive of the bundled appliance concept even if someone produces a VM - CentOS or another variation on the same theme. ..mike..

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