- sr-users - kamailio.org

[Users] Problem registering the UA with openSER(tls enabled)
by Ncheeku Baranov 03 Jan '07

03 Jan '07

Hi, I am trying to make my non-TLS/TLS UA register with my TLS enabled openSER. Currently I am just working on my local machine with the client UAs on the same subnet,(so there is only one domain, but its not named). Below is my configuration file: disable_tls = 0 listen = tls:10.30.100.41:5061 tls_verify_server = 1 tls_verify_client = 0 tls_require_client_certificate = 0 tls_method = TLSv1 tls_certificate = "/usr/local/etc/openser/tls/user/user- cert.pem" tls_private_key = "/usr/local/etc/openser/tls/user/user- privkey.pem" tls_ca_list = "usr/local/etc/openser/tls/user/user-calist.pem" However, with the above configuration the client UAs couldnot register and I got 408 Request Time out Message. Is there any field that is missing to make this simple scenario work? What should be the values of "tls_client_domain" and "tls_server_domain" fields in this case? I noticed that when I start the openSER without TLS support using "openserctl start" and do "ps -e" after that, there are more openSER processes running than if I start openSER with TLS support in which case I see very few of these processes running. Your help is much appreciated.... Best regards, NCheeku

3 11

[Users] 487 Transaction Terminated
by KOK FOON YEW 03 Jan '07

03 Jan '07

This is pertaining to call cancel operation where the calling UA sends a CANCEL request via openser to the called UA. After the UA sends ACK in response to 487 Transaction Terminated response, 487 responses are re-transmitted by openser to the UA as if the SIP proxy didn't regcognise the ACK request. I have tested with other UA and compare the differences between the ACK sent by UA that worked and UA which has this problem. The difference is that the ACK sent by the UA having this problem has a route header added. Foon-Yew Kok [This e-mail is confidential and may be priviledged. If you are not the intended recipient, please kindly notify us immediately and delete the message from your system; please do not copy or use it for any purpose, nor disclose its contents to any other person. Thank you.] ---ST Electronics Group---

2 1

[Users] Accounting using dialog and sst module
by Asterisk Expert 03 Jan '07

03 Jan '07

Hello, I am currently trying to do accurate billing with openser. I found that dialog and sst modules can be used to determine failed transactions. i.e. If UA crashed and no BYE received at all. I tried to configure dialog and sst modules as specified in the documentation. But active dialogs keep incrementing even after hangup. It would be nice if someone can post a good example. I am using cvs head of openser. Snippet of config is as below: ###################### DIALOG ###################### modparam("dialog", "dlg_flag", 13) modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "use_tight_match", 1) modparam("dialog", "enable_stats", 1) ###################### SST ###################### modparam("sst", "enable_stats", 1) modparam("sst", "timeout_avp", "$avp(i:10)") modparam("sst", "sst_flag", 14) route { ........ ........ if(method=="INVITE") { setflag(13); # set the dialog flag setflag(14); # Set the sst flag avp_write("15","$avp(i:10)"); }; if (method=="INVITE") { record_route_preset("192.168.1.2:5060;nat=yes;did=1"); }; ......... ......... } -- Regards Asterisk Expert

3 6

[Users] Help: how can I create one SIP account
by CESAR HUAMANI H. 03 Jan '07

03 Jan '07

Hello, Today I installed the openser in my debian etch since the CVS, I was looking information about how I can creat accounts SIP for my Endpoints (softphone and Hardphone ATA), please help me I'm new with the OPENSER Best regards

2 1

[Serusers] Detecting if username is in subscriber table
by Simon 02 Jan '07

02 Jan '07

Dear All, I would like to detect if a URI is for a user that is in the subscriber table. If I just use lookup("location") then I only detect those users that have registered. If a phone is switched off it is not registered and so will not be in the location table. However it will be in the subscriber table and I need to be able to check this from within the ser.cfg file to allow correct routing etc. Any Ideas ? Many Thanks Simon

2 1

[Users] Call forward record and Remove from callid in Openser
by Howard Tang 02 Jan '07

02 Jan '07

Hi All, I hope someone can help me with the following questions I have. 1. I have set up accounting module for record the call, simple outgoing call and incoming call are ok. However, if a sip phone activated the call forward, the call record doesn't show which sip client forward the call. e.g. A call B and B is the sip phone which forward the call to C. In the record, I can only see A call C. How can I see a record that B call C instead of A call C in the record? 2. I have connected to a PSTN gateway, and i have some internal sip users which can make a pstn call, the sip proxy will send their own private number a caller id to the pstn gateway. Since the callid is a invalid number, many Mobile phone networks provider reject the call. I want to remove the from callid when the call being send to the pstn gateway, how can i do this? If someone can send me the code that will be great. -- Howard Tang

2 1

[Users] Presence and Location
by Klaus Darilion 30 Dec '06

30 Dec '06

Hi! What kind of presence does the presence module support? Only the "basic" presence format PIDF (RFC 3863) or also PIDF extensions like the GEOPRIV Location Object Format RFC 4119? Or in other words - is the presence module aware of the XML payload or is it agnostic to the presence payload? regards klaus

2 1

[Users] totag_avpid
by Klaus Darilion 30 Dec '06

30 Dec '06

Hi! I just found this parameter in sl and presence module, but the usage is not described well. What is it's purpose? I guess that the presence module exports the totag into this AVP and the sl module use this totag in its response. I think it is not a good idea to have it configureable in 2 modules. We had the same problem with the "use_domain" parameter in several modules which was fixed recently. Maybe this can be fixed the same way - having it configurable in sl module, and presence module look ups the value on loading (or generate a error if not defined). Further, the presence module's parameter documentation lack the default values. regards klaus

2 1

[Users] Inspiring New Year wishes from Sunkara Ravi
by Sunkara Ravi 30 Dec '06

30 Dec '06

Dear Users, Sunkara Ravi [ sunkara.raviprakash.feb14(a)gmail.com ] has sent you an ecard from 123Greetings.com. Send free ecards from 123greetings.com with your choice of colors, words and music. Your ecard will be available with us for the next 30 days. If you wish to keep the ecard longer, you may save it on your computer or take a print. To view your ecard, choose from any of the following options: -------- OPTION 1 -------- Click on the following Internet address or copy & paste it into your browser's address box. http://www.123greetings.com/view/GX91230040549763 -------- OPTION 2 -------- Copy & paste the ecard number in the "View Your Card" box at http://www.123greetings.com Your ecard number is GX91230040549763 If you need help in viewing your card or any other assistance, please visit our Help / FAQ section located at http://www.123greetings.com/help/ If you need further help, feel free to write to us at support(a)123greetings.com Best wishes, Postmaster, 123Greetings.com *If you would like to send someone an ecard, you can do so at http://www.123greetings.com

1 0

Re: Re: [Users] how to configure failure route for b2bua or GW
by liu york 30 Dec '06

30 Dec '06

yes,the failure route works,but the client phone still receive 408 messeage. I try to use if(t_check_status(408|483)) drop; at failure route ,but faild :( ======= 2006-12-29 16:28:57 您在来信中写道：======= >Do no use forward(), just rewritehost and afterwards t_on_failure and t_relay. > >Samuel. > >2006/12/29, liu york <york(a)goldentek.biz>: >> hi all >> >> I try to build failure route to another backup b2bua or pstn gw,it seems can work after timeout ,but unfortunate,the openser send a 408 messeage to phone before failure route.,config like that,have any general config to failover or loadbalance for me.Many many thx. >> >> route[4] { >> >> log(1, "b2bua"); >> rewritehost("192.168.1.204"); >---> #forward("192.168.1.204:80"); >> t_on_failure("2"); >> t_relay(); >> } >> >> failure_route[2] { >> log(1, "b2bua"); >> rewritehost("192.168.1.226"); >---> #forward("192.168.1.226:5060"); >> t_relay(); >> } >> >> >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> > = = = = = = = = = = = = = = = = = = = = 　　　　　　　　致礼！　　　　　　　　york 　　　　　　　　york(a)goldentek.biz 　　　　　　　　　　2006-12-30

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[Serusers] Re: [Serdev] SIP Stack development
by Greger V. Teigre 29 Dec '06

29 Dec '06

Hi Alan, See inline. Alan Basinger wrote: > Thanks Greger, > I know ho hard it is to keep up with the documentation of a rapidly > changing project and also know what a challenge it is to manage the > code. I hope we can use the base of SER to create some new and > exciting applications uing SIP and add some usefull features and > functionality to the base code for everyone to benifit from. > > I appreciate the quick reply and answering a newbs questions as I may > have more as we move along. :-) Sure, I realize that lack of documentation can be replaced (partially and only temporarily) by responsive support on mailing lists. It is also a way to establish the actual scope of documentation work. > > Hi Alan, > I'm heading the new documentation effort. An updated list of the > supported rfcs and standards is on my list of needed docs... I'm on > mobile right now, so I > cannot search for you, but on iptel.org/listsearch you can search for > Janak and supported rfcs. I think he posted an update recently. > > As for the specific draft, I'm not familiar with it. In general > though, media is not handled by ser, only messaging. > The draft is for allowing messaging as well as media to be sent to a > loopback to mesure actual network conditions to and from the device > without actually setting up a call. Very usefull and we have many > vendors implimeting it in there CPE. (Linksys, Polycom, SNOM, > Panasonic, etc.) I woul like to see about developing this and many > other features / functionaility into SER. I worked with one of the > authors of this draft and it realy makes sense. Here is a link to it. > > http://www.ietf.org/internet-drafts/draft-ietf-mmusic-media-loopback-05.txt > Aha, I recognize the name now :-) Without knowing the details of the draft, I have the following comments: * In general you want your servers out of the rtp path (i.e. direct media between UAs), in fact, ser is never in the rtp path. However ser is very good at doing message processing, forwarding and all sorts of "tricks" * RTP may be handled by three separate software components: rtpproxy, mediaproxy, and sems (mediaproxy is not an iptel.org project, the others are). SER modules (nathelper and mediaproxy) are used to communicate with these external components (locally or over udp) and for mangling the SDP payload * Your measurements will probably need to follow the media path of a normal session. If not, you will not be able to measure for example real latency on your ex. rtpproxy server(s). SEMS is a very good general-purpose media server, while rtpproxy/mediaproxy are more dedicated for proxying of rtp for NAT-purposes etc * Hence, you probably need to implement sdp loopback intp the rtp handling component. SEMS may be the easiest way due to its very modular and powerful plugin interface (it's also a B2BUA). However, as sems don't stay in the media path except in conferences etc, you don't really measure "real world..." * I suspect you can create a ser module (or extend nathelper) that implements the basic loopback stuff using rtpproxy without touching rtpproxy (too much) > As for development, ser has a very efficient core exposing a module > interfaces. Modules can implement functions, parameters, avpairs (i.e. > set variables), and > selects (give access to certain info related to a message). > I found a copy of the 2002 developers guide and have been starting > there but an concerned that it may be too outdated any idea when a new > draft will be available for review? The basic concepts are still relevant. It has not been updated to cover to the attribute value pairs and the selects. We are currently focused on the user documentation for the next release, but we are debating what the next steps are. A how-to for module development is already on the list, possible with a retouch of the old developers guide. You can start out with one of the simpler modules (like textops or dispatcher) and modify to your needs. Understand the plugin interface, lumps and shared memory, and you are pretty much there ;-) > > Contributions are always welcome. We have just introduced a new > classification of modules, thus allowing experimental modules into the > cvs without too much > fuss. > Good to hear as me and my partner in crime and I hope to be able to > add some value with our ideas. :-) g-) > > BTW, start with ser ottendorf. It is getting close to release and has > many new important improvements. > > Feel free to ask questions on this list. > g-) > > Thanks again > > Alan Basinger > > ------- Original message ------- > From: Alan Basinger <droidgeneral(a)yahoo.com> > Sent: 27.12.'06, 12:43 > > > Hello all, > > I definitely am a newby with SER but not with SIP and I am trying to > find out where the supported SIP methods are? > > Specifically how would I go about finding out if SER supports the > Media Loopback draft or other ratified or non ratified components. > > Also if not supported where would I being the coder that I am not be > able to modify the code myself to add these functions and then > distribute them back to > the community? > > > > Thanks in advance > > > > Alan > > > > __________________________________________________ > > Do You Yahoo!? > > Tired of spam? Yahoo! Mail has the best spam protection around > > http://mail.yahoo.com <http://mail.yahoo.com/> > > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com

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[Users] New Year's Eve invitation from Sunkara Ravi
by Sunkara Ravi 29 Dec '06

29 Dec '06

Dear Users, Sunkara Ravi [ sunkara.raviprakash.feb14(a)gmail.com ] has sent you an ecard from 123Greetings.com. Send free ecards from 123Greetings.com with your choice of colors, words and music. Your ecard will be available with us for the next 30 days. If you wish to keep the ecard longer, you may save it on your computer or take a print. To view your ecard, choose from any of the following options: -------- OPTION 1 -------- Click on the following Internet address or copy & paste it into your browser's address box. http://www.123greetings.com/view/HW31229040825964 -------- OPTION 2 -------- Copy & paste the ecard number in the "View Your Card" box at http://www.123greetings.com Your ecard number is HW31229040825964 If you need help in viewing your ecard or any other assistance, please visit our Help/ FAQ section located at http://www.123greetings.com/help/ If you need further help, feel free to write to us at support(a)123greetings.com Best wishes, Postmaster, 123Greetings.com *If you would like to send someone an ecard, you can do so at http://www.123greetings.com

1 0

[Users] Cheerful New Year wishes from Sunkara Ravi
by Sunkara Ravi 29 Dec '06

29 Dec '06

Dear Users, Sunkara Ravi [ sunkara.raviprakash.feb14(a)gmail.com ] has sent you an ecard from 123Greetings.com. Send free ecards from 123Greetings.com with your choice of colors, words and music. Your ecard will be available with us for the next 30 days. If you wish to keep the ecard longer, you may save it on your computer or take a print. To view your ecard, choose from any of the following options: -------- OPTION 1 -------- Click on the following Internet address or copy & paste it into your browser's address box. http://www.123greetings.com/view/HW41229040814251 -------- OPTION 2 -------- Copy & paste the ecard number in the "View Your Card" box at http://www.123greetings.com Your ecard number is HW41229040814251 If you need help in viewing your ecard or any other assistance, please visit our Help/ FAQ section located at http://www.123greetings.com/help/ If you need further help, feel free to write to us at support(a)123greetings.com Best wishes, Postmaster, 123Greetings.com *If you would like to send someone an ecard, you can do so at http://www.123greetings.com

1 0

[Users] how to configure failure route for b2bua or GW
by liu york 29 Dec '06

29 Dec '06

hi all I try to build failure route to another backup b2bua or pstn gw,it seems can work after timeout ,but unfortunate,the openser send a 408 messeage to phone before failure route.,config like that,have any general config to failover or loadbalance for me.Many many thx. route[4] { log(1, "b2bua"); rewritehost("192.168.1.204"); forward("192.168.1.204:80"); t_on_failure("2"); t_relay(); } failure_route[2] { log(1, "b2bua"); rewritehost("192.168.1.226"); forward("192.168.1.226:5060"); t_relay(); }

2 1

Re: Re: [Serusers] Creating particular mysql schema with the given db user name - bad db connection url
by "tzieleniewski" 28 Dec '06

28 Dec '06

worked but another problem apeared:) the script has created the ser database user and the database ser_0_9_7 for this user Ive changed the database connection urls to the following: fifo_db_url="mysql://ser:heslo@localhost/ser0_9_7" modparam("auth_db|uri_db|usrloc|domain", "db_url", "mysql://ser:heslo@localhost/ser0_9_7") But despite that ser try too connect to the "ser" database: stateless - initializing 0(0) Maxfwd module- initializing textops - initializing 0(0) AVPops - initializing acc - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 109568 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 1(0) INFO: fifo process starting: 3499 1(0) new_connection(): Access denied for user 'ser'@'localhost' to database 'ser' 1(0) db_init(): Could not create a connection 1(0) ERROR: acc_db_init: unable to connect to the database 1(0) init_mod_child(): Error while initializing module acc 1(0) ERROR: open_uac_fifo: init_child failed 1(0) starting fifo server failed 1(0) INFO: signal 15 received > On Thu, Dec 28, 2006 at 02:14:18PM +0100, TZieleniewski wrote: > > > Ok but I generalny dont want to install ser I just want to compile it > > and then have two different instances of ser on the same machine > > (different releases)<br> > > gen_ha1.c and gen_ha1.d are in utils/gen_ha1 directory , there is a > > makefile included will it work if I will invoke make in that directory?<br> > > yep, it will works. Just compile and change the path to gen_ha1 in > ser_mysql.sh. > > > > <br> > > Tomasz<br> > > </body> > > </html> > > -- > Alexandr Dubovikov * baron@iRC RusNet * mailto:shurik@start4.info > AD1-UANIC * ICQ: 122351182 * http://www.start4.info

2 3

[Serusers] Radius Accounting Help
by Arun Kumar 28 Dec '06

28 Dec '06

Hi, I've configured my radius + mysql and openser. Now openser is sending accounting packet to radius server and I'm able to see accounting packets coming to my radius server but the problem is : Its sending the Acct-Status-Type = Start Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = Ack packet twice and then it shows rlm_radutmp: Login entry for NAS august port 5060 duplicate and then two stop packet: Acct-Status-Type = Stop Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = Bye and then it shows rlm_sql (sql) in sql_accounting: stop packet with zero session length and after second stop packet it shows rlm_radutmp: Logout for NAS august port 5060, but no Login record please help. thanks arun

2 4

[Users] lock error with configuration mistake
by Jean-François SMIGIELSKI 28 Dec '06

28 Dec '06

I wrote this message a few days ago: http://openser.org/pipermail/devel/2006-December/005103.html I am sorry for the lack of informations I gave, I could investigate a little bit about it. Here is the full context: When an INVITE comes, I dispatch and t_relay it to a server. If the transaction makes an error, I wrote in the configuration to manage this whith a failure route. In this failure route, I dispatch and t_relay the transaction to a server for an other service. And then appears an strange error : * with an openser compiled with the default flags, the UDP that received the last UDP packet infinitely loops on the "sched_yield" function, in the "tsl" function in fastlock.h. * with an openser compiled with flags that make him use pthread for locking purposes, the UDP process hangs, while trying to get a mutex. It appears I forgot a call to append_branch in the failure route. Maybe the process tried to get the same lock twice. Is it known for OpenSER 1.1.0? If it is, is it corrected in later stable releases ? Thanks a lot for your answer! JF Smigielski. ________________________________________________________________________ iBELGIQUE, exprimez-vous ! http://web.ibelgique.com/

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[Serusers] Presence support: PUBLISH/SUBSCRIBE
by raj gopal 28 Dec '06

28 Dec '06

Hi, I have installed ser-0.10.99-dev35-pa-4.1 and following up presence handbook. i have configured XCAP using apache. i am getting following errors with ser.cfg of ser package. For PUBLISH request : 0(13411) get_pres_uri(): Error while extracting plain URI 0(13411) handle_publish(): Error while extracting presentity URI For SUBSCRIBE request: 0(13411) ERROR: rls_handler.c:484: XCAP query problems 0(13411) XCAP query failed 0(13411) ERROR: rls_handler.c:484: XCAP query problems 0(13411) XCAP query failed have any one succeed with presence? help me... *-Venkat *

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[Serusers] AVPops and Textops to chang username of request uri
by chungyu 28 Dec '06

28 Dec '06

Hi: I use module AVPops and Textops do chang username of request uri. example: john(a)10.10.16.23 I hope chang => 4762(a)10.10.10.22 my configuration sample: if (uri=~"john@") { avp_write("4762", "s:phone"); subst_user(/(.*)/$avp(phone)/'); rewritehostport("10.10.10.22:5060"); forward(uri:host, uri:port); avp_delete("s:phone"); }; but reply is "Call failed:Device not found" I try this sample is OK if (uri=~"john@") { avp_write("4762", "s:phone"); subst_user(/(.*)/4762/'); rewritehostport("10.10.10.22:5060"); forward(uri:host, uri:port); avp_delete("s:phone"); }; OS: Freebsd 5.4 SIP server: SER-0.9.6 please tell my problem is on? Thank you Chungyu This is my Configuration: # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 #children=4 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/avp.so" #loadmodule "/usr/local/lib/ser/modules/uri.so" #loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==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("xx.xx.xx.xx", "subscriber")) { www_challenge("xx.xx.xx.xx", "0"); break; }; save("location"); break; }; if (uri=~"john@") { avp_write("4762","phone"); subst_user('/(.*)/$avp(phone)/'); rewritehostport("10.10.10.22:5060"); forward(uri:host, uri:port); avp_delete("phone"); }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; }

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[Users] MP-104 FXO issues
by Nick De Cristofaro 27 Dec '06

27 Dec '06

Can't make outbound calls, though not related directly to OpenSER.. Maybe someone here knows how to configure these guys. I have inbound calls being automatically dialed into a * extension but If I make any outbound calls I get a syslog output: 16:45:21.609 : 192.168.1.211 : WARNING : ( lgr_call)(458 ) !! [ERROR] Call::GetEndPoint- Can't find endpoint for phone number xxxxxx2018 16:45:21.593 : 192.168.1.211 : WARNING : ( lgr_psbrdif)(457 ) !! [ERROR] AcBoard::GetEndPoint- Can't find EndPoint for Dest:xxxxxxx2018 Source:xxxxxxx11526 SourceIp:c0a80188 16:45:21.593 : 192.168.1.211 : WARNING : ( lgr_TrnkGrp)(456 ) !! [ERROR] #0:TrunkGroup::AllocateEndPoint- Can't find EndPoint for phone number xxxxxx2018 16:45:21.593 : 192.168.1.211 : NOTICE : ( lgr_stk_ses)(455 ) SIPStackSession::HandleStackSetupEV - NEWCALL: SrcPN=0 Any help would be great. Also, caller ID doesn't work - only shows the caller ID of whatever is set at the endpoint phone number settings and not what is actually coming on the line :( Nick

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[Users] TLS problem, not able to start openSER
by Ncheeku Baranov 27 Dec '06

27 Dec '06

Hi, I just compiled openSER with TLS support. I checked that TLS = 1 in the Makefile when I compiled openSER. Now when I try to uncomment the parameters in the openser.cfg to enable the TLS support and restart openSER it does not start (I am using openserctl start command to start openser). It gives an error saying ERROR:PID file /var/run/openser.pid does not exist -- OpenSER start failed. I am using the following parameters in the openser.cfg file for the TLS support: disable_tls = 0 listen = tls:10.30.100.41:5061 tls_verify = 1 tls_require_certificate = 0 tls_method = TLSv1 tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" tls_ca_list = "usr/local/etc/openser/tls/user/user-calist.pem" I have checked that all the paths are correct in defining the tls_certificate, tls_private_key and tls_ca_list. I used the source tarball openser-1.1.0-tls_src.tar.gz<http://www.openser.org/pub/openser/latest/src/openser-1.1.0-tls_src.tar.gz> for installing the openser. Your help is much appreciated. Thanks NCheeku

2 2

[Users] SER vs openSER
by pankaj.munjal＠wipro.com 27 Dec '06

27 Dec '06

Hi all, I am a SER/openSER newbie and was looking at an option of using these SIP stacks. Anyone having any doc/link that compares these two SIP servers. The major criteria that we are looking for are: - Simplicity of using the SIP stack (Most IMP) - Perfomance - Support for Presence Any pointers in this direction will be highly appreciated. Thanks, Pankaj The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

2 1

[Users] subst_body with nated client problem
by Oleg A Chernov 27 Dec '06

27 Dec '06

Hello everyone! My setup is as follow: for local PSTN calls UA (with G.711 and G.729) - my Openser - my Asterisk B2BUA for prepaid - PSTN gateway (G.711 codec only) for international calls UA (with G.711 and G.729) - my Openser - my Asterisk B2BUA for prepaid - ITSP gateway (G.729 codec only) B2BUA on the second leg (B2BUA - gateway) always negotiate the right codec, so no problem, but on the first leg (UA - B2BUA) it always negotiate codec that is more prefered by the client UA side. So we can get situation, when the codecs are different on the both legs. Inspired by the http://lists.iptel.org/pipermail/serusers/2006-August/030129.html I decided to normalize codec preference. Sorry for such a long introduction, at last here goes my problem. I have the following code in my openser.cfg in INVITE section # INVITE section route[3] { ... if ( route to G.711 capable gateway) { subst_body("/^m=audio(.*)RTP\/AVP(.*) 8 (.*)/m=audio\1RTP\/AVP 8 \2 \3/ig"); .... } ... } and it works like a charm for non-nated clients with real IPs. But when we should fire use_media_proxy() for a nated one client we get the problem. SDP body before our substitution (UA-Openser): v=0 o=20000 12893 12893 IN IP4 192.168.2.134 s=ATA186 Call c=IN IP4 192.168.2.134 t=0 0 m=audio 16384 RTP/AVP 0 18 8 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=yes a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 After (Openser - B2BUA): v=0 o=20000 12893 12893 IN IP4 192.168.2.134 s=ATA186 Call c=IN IP4 217.77.208.184 t=0 0 m=audio 16384 RTP/AVP 8 0 18 101\01560136\012a=rtpmap:0 PCMU/8000/1 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=yes a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 So our subst_body() rearrange codecs as expected, but use_media_proxy() (it her fault, I think) inserted the mediaproxy fixed port number 60136 between symbols with ASCII codes \015 CR and \012 LF. This behavior was accepted by stable Openser 1.1 and latest cvs 1.2. So the question is : What to do and who is guilty ? :-) -- Best regards, Oleg A. Chernov OAC4-RIPE WildPark ISP, Nikolaev, Ukraine +380512 470555 +380512 500314

2 1

[Serusers] SER server as a client
by jk 27 Dec '06

27 Dec '06

Hello List, I am working on an assignment to develop a load balancing asterisk cluster. I started with google and found out that SER+ASTERISK is the best way to do it. Here is what I am trying to do. I have a sip account / DID (18xx xxx xxxx ) from an abc company. Now the company is gateway for between my asterisk server and PSTN network. My Asterisk server registers with abc company's gateway to receive and make calls. How can I do load balancing in this case. I understand that my SER server should register with the gateway, the gateway should forward call to SER server and then the server should forward call to one of the asterisk servers. Is it possible have the SER server register with another SER or Asterisk server and act like a client to receive and make calls? Can some one please point me to some documentation or case study? Thank you, -JK

2 2

[Users] LCR db_only mode problem
by Jayesh Nambiar 26 Dec '06

26 Dec '06

Hi all, A merry christmas and a great new year to all of you. I am trying to implement LCR in my openser 1.1. My problem is that I have around 40,000 prefix patterns which i need to enter into the LCR table. When I populate my LCR table and try to start my openser it dies. I can see the following messages in my log: convert_row: No memory left convert_rows: Error while converting row #9055 convert_result: Error while converting rows store_result: Error while converting result lcr_reload_gws(): Failed to query lcr data ERROR: lcr:mod_init(): failed to reload gateways and routes init_mod(): Error while initializing module lcr The openser starts properly when I tested with 1000 rows in the LCR table but with 40,000 rows it just dies. Can anyone please help me overcome this issue. Is there any parameter available to make openser properly run with these many rows in the LCR table. I am using the LCR in db_only mode. I have not tried this in caching mode. Will the caching mode work with these many rows?? w/regards, jayesh. Send free SMS to your Friends on Mobile from your Yahoo! Messenger. Download Now! http://messenger.yahoo.com/download.php

2 1

[Serusers] Flex and Bison
by Digital Tuffy 26 Dec '06

26 Dec '06

Could anybody pls brief the use of Flex and Bison with SER. _________________________________________________________________ Christmas is drawing near and the excitement is in the air! Check out this cool photofeature on Christmas celebrations the world over http://www.msnspecials.in/photogallery/photogallery.asp?id=a200611814137906

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[Users] Sending REGISTER adn INVITE--plz help
by imran ahmed 26 Dec '06

26 Dec '06

Hi Users; I am new to use openser so having some problems. I have installed openser-1.0.0.I have the task to do load test of openser through SIPp with AUTHENTICATION. I have added some users in the subscriber table of openser. Now the problem is how to set the openser.cfg so that it can response for INVITE and REGISTER request both like: The flow goes like: 1)SIPp UAC send REGISTER -------> OPENSER 2)407 Proxy Authentication required <------ OPENSER 3)SIPp send REGISTERwith credential------>OPENSER(check from subscrible tables and on the bases of that allow or disallow) hope someone help me; Thanks and Regards; Imran __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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Re: [Users] OpenSer Sending CANCEL after receiving 200 OK for a INVITE
by Klaus Darilion 26 Dec '06

26 Dec '06

Hi! TAke a look at this line: Dec 20 14:47:46 openser[17366]: oRnbgCwOGh9EngQ1(a)xx.88.13.xx request's destination set: $ds=Contact: sip:02111@xx.87.220.xx:6060, sip:02111@xx.87.220.xx:6060 The destination set include 2 destinations: sip:02111@xx.87.220.xx:6060 and sip:02111@xx.87.220.xx:6060 Thus, openser will fork the call with 2 branches (if you take a look at the message flow you will see 2 INVITEs from openser to Asterisk which are nearly identical, only the branch parameter in openser's Via header will be different.) Asterisk is buggy and does not recognize that the 2 INVITEs are different branches but think the second INVITE is a retransmission. Thus when Asterisk accepts the call (accepting on of the two branches), openser will cancel the other branch. As the second branch is identical to the first one, it will send the CANCEL to Asterisk. As Asterisk cann not handle multiple dialog it wil think this is the recently accepted call and will hang up. Conclusion: fix your openser config. Probably in your exec script (or somewhere in openser.cfg) you create a second branch which will be filled with the same URI as the first destination. regards klaus PS: please always Cc the mailing list. Md. Samiul Aftad Chowdhury wrote: > Sorry for quite a late reply... > > but i found same scenario when i am sending the call to an asterisk box. > > Here is the log you asked for: > > In following scenario: > > xx.35.32.xx <- My Openser IP which is running on 5060 port > > xx.87.220.xx:6060 <- Asterisk Box where i am forwarding the calls > which is running on 6060 port > > xx.88.13.xx <- ipphone ip > > Same thing is happening here just after receiving 200 OK Openser > generating a CANCEL. > > i am using : > > ============================================ > exec_dset("/usr/local/route"); # here route is a script which returns > a desired host port (URI) > append_branch(); > ============================================ > > to rewrite the host port. > > > Dec 20 14:47:45 openser[17368]: Invite -> > [CallerID=sip:707234121839@xx.35.32.xx] and [DialedNo=02111] > Dec 20 14:47:46 openser[17366]: Invite -> > [CallerID=sip:707234121839@xx.35.32.xx] and [DialedNo=02111] > > Dec 20 14:47:46 openser[17366]: oRnbgCwOGh9EngQ1(a)xx.88.13.xx request's > first branch: $br=sip:02111@xx.87.220.xx:6060 > Dec 20 14:47:46 openser[17366]: oRnbgCwOGh9EngQ1(a)xx.88.13.xx request's > all branches: $bR=sip:02111@xx.87.220.xx:6060 > Dec 20 14:47:46 openser[17366]: oRnbgCwOGh9EngQ1(a)xx.88.13.xx request's > destination set: $ds=Contact: sip:02111@xx.87.220.xx:6060, > sip:02111@xx.87.220.xx:6060 > Dec 20 14:47:46 openser[17366]: oRnbgCwOGh9EngQ1(a)xx.88.13.xx request's > destination uri: $du= > Dec 20 14:47:46 openser[17366]: oRnbgCwOGh9EngQ1(a)xx.88.13.xx request's > uri: $ru=sip:02111@xx.87.220.xx:6060 > > Dec 20 14:48:01 openser[17366]: oRnbgCwOGh9EngQ1(a)xx.88.13.xx request's > first branch: $br=<null> > Dec 20 14:48:01 openser[17366]: oRnbgCwOGh9EngQ1(a)xx.88.13.xx request's > all branches: $bR= > Dec 20 14:48:01 openser[17366]: oRnbgCwOGh9EngQ1(a)xx.88.13.xx request's > destination set: $ds=<null> > Dec 20 14:48:01 openser[17366]: oRnbgCwOGh9EngQ1(a)xx.88.13.xx request's > destination uri: $du= > Dec 20 14:48:01 openser[17366]: oRnbgCwOGh9EngQ1(a)xx.88.13.xx request's > uri: $ru=sip:02111@xx.87.220.xx:6060 > > > > On 12/7/06, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: >> >> Hi! >> >> I could not find any suspect messages. I guess there is some parallel >> forking inside openser, thus, the second branch is cancelled when the >> first branch picks up. >> >> Can you add this just before t_relay(): >> >> xlog("L_INFO","$ci request's first branch: $$br=$br\n"); >> xlog("L_INFO","$ci request's all branches: $$bR=$bR\n"); >> xlog("L_INFO","$ci request's destination set: $$ds=$ds\n"); >> xlog("L_INFO","$ci request's destination uri: $$du=$du\n"); >> xlog("L_INFO","$ci request's uri: $$ru=$ru\n"); >> >> >> regards >> klaus >> -- Klaus Darilion nic.at

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[Users] Merry Christmas and A Happy New Year!
by Daniel-Constantin Mierla 24 Dec '06

24 Dec '06

Hello, it was a year full of achievements and events for OpenSER. We had a major release in summer (1.1.0), and a continuous increase in features set and robustness. What was new in 1.1.0 was summarized in release news: http://www.openser.org/index.php?option=com_content&task=view&id=47&Itemid=… Since then we went though many milestones. We had the first OpenSER summit where we were able to meet many folks from the lists, have nice chats and present interesting VoIP solutions based on OpenSER. http://www.openser.org/index.php?option=com_content&task=view&id=57&Itemid=… A very important statistic is the number of new modules and features introduced in current development version after 1.1.0: - domainpolicy - policies to connect federations - imc - instant messaging conferencing - mi_fifo and mi_xmlrpc - FIFO and XMLRPC transports for the new management interface (MI) - perl - embed perl programming in configuration file - presence - SIMPLE Presence Server implementation - pua, pua_mi, pua_usrloc - presence user agent client implementations for user location records and management interface - seas - connector to SIP Application Server - WeSIP - Java SIP Servlet Application Server (http://www.wesip.eu) - snmpstats - SNMP (Simple Network Management Interface) interface to OpenSER statistics - sst - SIP session timer support - xmpp - transparent SIP-XMPP gateway Documentation and news about all these: http://www.openser.org/docs/modules/1.2.x/ http://www.openser.org/index.php?option=com_content&task=blogsection&id=2&I… A lot of other significant improvements and addons were included in existing code (management interface, fetch database support, postgres, usrloc, dialog, enum, acc modules polishing), openser web administrator was developed. There is not very long time to the next release, along with few pending new features, the next period will be allocated to core components like timers, dns and configuration file flexibility. By end of January we should approach a new testing phase. This year the OpenSER community collaboration exploded, dokuwiki has lot of content, many tutorials were submitted, new developers and many modules sponsored by companies, mailing lists and web forum are places to get reliable information about OpenSER... We want to thank to all of you (developers, contributors with patches, tools and documentation, testers and OpenSER users) for supporting the project. 2006 was a very good year for OpenSER and 2007 looks very challenging. We wish you Merry Christmas and a very Happy New Year!

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[Serusers] Service Empty
by raj gopal 23 Dec '06

23 Dec '06

Hi All, I am trying presence snapshot SER. I am getting 0(4189) Empty service! 0(4189) ERROR: rls_handler.c:484: XCAP query problems 0(4189) XCAP query failed error for SUBSCRIBE request. I am following presence handbook. Whats worng i am doing????? Can anyone help me?? Thanks in advance Venkat

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[Serusers] Re: [asterisk-users] Happy X-mas
by Arun Kumar 23 Dec '06

23 Dec '06

Hi All, Wish you a very HAPPY and Merry Christmas to all and your beloved once. Arun On 12/23/06, Josué Conti <josueconti(a)gmail.com> wrote: > > Hi ALL, > > ** I like very to desire you and your family, a Merry Christmas, with much > love, peace, professional and personal success. > > Best Regards > Josue > > > 2006/12/23, raviprakash sunkara <sunkara.raviprakash.feb14(a)gmail.com>: > > > > > > Hello > > * * * * * * * * > > * * * * * Happy X-mas and Adv Happy New Year ....... > > ************************************ > > **** > > **** > > ******** > > ** > > > > ************* > > > > > > -- > > Thanks and Regards > > Ravi Prakash Sunkara > > ravi.sunkara(a)hyperion-tech.com > > M:+91 9985077535 > > O:+91 40 23114549 > > F:+91 40 40208727 > > ravi.sunkara(a)hyperion-tech.com > > www.hyperion-tech.com > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com<http://easynews.com/>-- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >

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[Users] Happy X-mas
by raviprakash sunkara 23 Dec '06

23 Dec '06

Hello * * * * * * * * * * * * * Happy X-mas and Adv Happy New Year ....... ************************************ **** **** ******** ** ************* -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

1 0

[Users] Re: [Devel] New module for OpenSER - Application Server interconnection
by Bogdan-Andrei Iancu 22 Dec '06

22 Dec '06

Hi Elias, (also fwd. to "users" to let everybody know) thanks for the contribution. I think that the new SEAS and PERL modules will give a lot more options when comes to integrate OpenSER with other application and what is more important , they will allow non-C extensions :) !! I will generate the docs and place them on the web also. BTW, are some default application that I can give a fast try? I would like to test some real-case apps to get the feeling. Regards and Merry Christmas, Bogdan Elias Baixas wrote: > Hello everybody, > > I just submitted a new module, called SEAS (which stands for Sip > Express Application Server). > This module allows a third party to create an Application Server and > connect it to OpenSER to interoperate seamlessly. > This permits the Application Server to trigger actions on OpenSER, > such as sending out a request, or replying a UAS transaction, in some > way similarly to what can be achieved using fifo, or MI commands > "t_uac_dlg" and "t_reply", but SEAS provides a much richer API to this > kind of tasks. > The module also allows OpenSER to notify the Application Server when > requests/responses come in, and then transfer the control over the > incoming request entirely to the Application Server. > Furthermore, the module implements a binary protocol that transfers > the SIP message along with information abouth the structure of the > message's headers (ie. where each header starts, and how long it is), > so the Application Server doesn't need to reparse the entire SIP > message, thus improving Application Server's performance very much. > The SEAS protocol also carries information about the transaction being > processed at OpenSER for every message, so that the Application Server > doesn't need to implement all the SIP-transaction machinery, thus > actually profitting from the highly optimized TM module at OpenSER. > > The Application Server can be programmed in any language, because the > SEAS module comunicates with it through a TCP socket. > > At the moment, SEAS module has been written to work with an > Application Server developed by VozTelecom. It is called WeSIP and can > be downloaded freely from http://www.wesip.eu . WeSIP is an > implementation of the SipServlet and HttpServlet standards, so it > allows a programmer to easily implement Web-Sip converged applications > (such as Click2call). > WeSIP is programmed entirely in Java, so you have all the tools and > facilities from Java and J2EE available to program SIP applications, > while already using OpenSER to process the SIP messages and > transactions at the low level, thus profitting both from the high > performance of OpenSER, and the high-level SIP-Application programming > framework which is SipServlet and the JAVA languaje. > > I hope this module will help create new usage scenarios for OpenSER, > allowing it to provide high-level Telephony Applications to users. > > The module is still in a alpha stage, but should be able for > production shortly. > > > best regards, > > Elias Baixas > > VozTelecom, Sistemas > http://www.wesip.eu > http://www.voztele.com > > _______________________________________________ > Devel mailing list > Devel(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/devel >

3 3

[Serusers] Invite going to my gateway
by Bill 22 Dec '06

22 Dec '06

Scenario: I have two test phones setup to a SER server. The two phones are 192.168.2.133 and 192.168.2.101. My server is 192.168.2.1. I can call from .101 to .133 fine. When I try to place a call from .133 this is what I get. 0(2734) SIP Request: 0(2734) method: <INVITE> 0(2734) uri: <sip:1001@192.168.2.1> 0(2734) version: <SIP/2.0> 0(2734) parse_headers: flags=1 0(2734) Found param type 232, <branch> = <z9hG4bK7e2902f2a>; state=16 0(2734) end of header reached, state=5 0(2734) parse_headers: Via found, flags=1 0(2734) parse_headers: this is the first via 0(2734) After parse_msg... 0(2734) preparing to run routing scripts... 0(2734) parse_headers: flags=128 0(2734) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(2734) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(2734) parse_headers: flags=8 0(2734) DEBUG: get_hdr_body : content_length=586 0(2734) end of header reached, state=9 0(2734) DEBUG: get_hdr_field: <To> [29]; uri=[sip:1001@192.168.2.1] 0(2734) DEBUG: to body [1001 <sip:1001@192.168.2.1> ] 0(2734) DEBUG: add_param: tag=4d4e374f5ebe6ab 0(2734) end of header reached, state=29 0(2734) parse_headers: flags=256 0(2734) get_hdr_field: cseq <CSeq>: <1338505295> <INVITE> 0(2734) is_preloaded: Yes 0(2734) grep_sock_info - checking if host==us: 13==13 && [192.168.2.136] == [192.168.2.136] 0(2734) grep_sock_info - checking if port 5060 matches port 5060 0(2734) after_loose: Topmost route URI: 'sip:192.168.2.136:5060;lr' is me 0(2734) parse_headers: flags=256 0(2734) found end of header 0(2734) find_next_route: No next Route HF found 0(2734) after_loose: No next URI found 0(2734) grep_sock_info - checking if host==us: 11==13 && [192.168.2.1] == [192.168.2.136] 0(2734) grep_sock_info - checking if port 5060 matches port 5060 0(2734) grep_sock_info - checking if host==us: 11==13 && [192.168.2.1] == [192.168.2.136] 0(2734) grep_sock_info - checking if port 5060 matches port 5060 0(2734) check_self: host != me 0(2734) DEBUG: t_newtran: msg id=45 , global msg id=43 , T on entrance=0xffffffff 0(2734) parse_headers: flags=-1 0(2734) parse_headers: flags=60 0(2734) t_lookup_request: start searching: hash=15747, isACK=0 0(2734) DEBUG: RFC3261 transaction matching failed 0(2734) DEBUG: t_lookup_request: no transaction found 0(2734) SER: new INVITE 0(2734) parse_headers: flags=-1 0(2734) check_via_address(192.168.2.133, 192.168.2.133, 0) 0(2734) WARNING:vqm_resize: resize(0) called 0(2734) DEBUG: reply sent out. buf=0x8104e98: SIP/2.0 1..., shmem=0xb6145728: SIP/2.0 1 0(2734) DEBUG: _reply_light: finished 0(2734) DEBUG: mk_proxy: doing DNS lookup... 0(2734) check_via_address(192.168.2.133, 192.168.2.133, 0) 0(2734) DEBUG: add_to_tail_of_timer[4]: 0xb6143eb8 0(2734) DEBUG: add_to_tail_of_timer[0]: 0xb6143ec8 0(2734) SER: new transaction fwd'ed 0(2734) DEBUG:destroy_avp_list: destroying list (nil) 0(2734) receive_msg: cleaning up 1(2737) DEBUG: timer routine:4,tl=0xb6143eb8 next=(nil) 1(2737) DEBUG: retransmission_handler : request resending (t=0xb6143da0, INVITE si ... ) 1(2737) DEBUG: add_to_tail_of_timer[5]: 0xb6143eb8 1(2737) DEBUG: retransmission_handler : done 0(2734) SIP Request: 0(2734) method: <CANCEL> 0(2734) uri: <sip:1001@192.168.2.1> 0(2734) version: <SIP/2.0> 0(2734) parse_headers: flags=1 0(2734) Found param type 232, <branch> = <z9hG4bK7e2902f2a>; state=16 0(2734) end of header reached, state=5 0(2734) parse_headers: Via found, flags=1 0(2734) parse_headers: this is the first via 0(2734) After parse_msg... 0(2734) preparing to run routing scripts... 0(2734) parse_headers: flags=128 0(2734) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(2734) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(2734) parse_headers: flags=8 0(2734) DEBUG: get_hdr_body : content_length=0 0(2734) end of header reached, state=9 0(2734) DEBUG: get_hdr_field: <To> [29]; uri=[sip:1001@192.168.2.1] 0(2734) DEBUG: to body [1001 <sip:1001@192.168.2.1> ] 0(2734) DEBUG: add_param: tag=4d4e374f5ebe6ab 0(2734) end of header reached, state=29 0(2734) parse_headers: flags=256 0(2734) get_hdr_field: cseq <CSeq>: <1338505295> <CANCEL> 0(2734) is_preloaded: Yes 0(2734) grep_sock_info - checking if host==us: 13==13 && [192.168.2.136] == [192.168.2.136] 0(2734) grep_sock_info - checking if port 5060 matches port 5060 0(2734) after_loose: Topmost route URI: 'sip:192.168.2.136:5060;lr' is me 0(2734) parse_headers: flags=256 0(2734) found end of header 0(2734) find_next_route: No next Route HF found 0(2734) after_loose: No next URI found 0(2734) grep_sock_info - checking if host==us: 11==13 && [192.168.2.1] == [192.168.2.136] 0(2734) grep_sock_info - checking if port 5060 matches port 5060 0(2734) grep_sock_info - checking if host==us: 11==13 && [192.168.2.1] == [192.168.2.136] 0(2734) grep_sock_info - checking if port 5060 matches port 5060 0(2734) check_self: host != me 0(2734) DEBUG: t_newtran: msg id=46 , global msg id=45 , T on entrance=0xffffffff Bill Hissong

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[Users] re: how to pass the Caller ID into uac_replace_from function ?
by don lin 22 Dec '06

22 Dec '06

Hi there, I saw all of you discussed about the postgres and I do have a question which somewhat related to postgres as well (please refer to the end of this email for detail). My question is that I have Caller ID (regular phone number) mapping to a SIP account stored in postgres, when I call from SIP device to a PSTN phone, telco carrier does not accept the SIP From field and requested to use regular phone number. I use exec_msg to call the external C program which read the Caller ID mapping to SIP account. However, I do not know how to pass this Caller ID into uac_replace_from. Were the postgres version work for this purpose ? Please advise if you have any suggestion. Thanks in advance, Don ------------------------------ Message: 5 Date: Thu, 21 Dec 2006 12:20:47 +0200 From: Daniel-Constantin Mierla <daniel(a)voice-system.ro> Subject: Re: [Users] postgres To: Klaus Darilion <klaus.mailinglists(a)pernau.at> Cc: users(a)openser.org Message-ID: <458A5FFF.4060003(a)voice-system.ro> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hello Klaus, On 12/21/06 12:05, Klaus Darilion wrote: > Hi Daniel! > > I've checked the postgres script: rel_1_1_0 works fine, but there is a > problem in unstable: The domain for the default admin user is asked > not for serweb tables but for the plain openser tables as well. Also > the subscriber table includes all the serweb columns even if serweb > tables are not installed. I think a merge with mysql version, with an option to say the underlaying DB engine is the right solution, to be easy to maintain -- same as we did with openserctl. I will have that in mind, but the best will be to register it to tracker. Cheers, Daniel > > regards > klaus > > > Daniel-Constantin Mierla wrote: >> Hello, >> >> On 12/20/06 19:36, Mark Price wrote: >>> >>> >>> On 12/20/06, *Klaus Darilion* < klaus.mailinglists(a)pernau.at >>> <mailto:klaus.mailinglists@pernau.at>> wrote: >>> >>> Hi Mark! >>> >>> Postgres should work well - I use it since ser 0.8. Just make sure >>> that >>> the hard disk does not get full, because this breaks the index >>> inside >>> postgres and postgres is getting real slow (re-create the index >>> if it >>> happens). >>> >>> Last time I tested openser_postgres.sh it worked fine. If you find >>> a bug >>> please let us know. >>> >>> Please check to use the latest versions from CVS (for 1.1 use CVS >>> rel_1_1_0) >>> >>> >>> The latest version from cvs doesn't work out of the box with openser >>> build from the release tarball, because the release tarball doesn't >>> included /usr/sbin/openser_gen_ha1 (although I could get past this >>> by replacing it with the md5sum equivelant). >>> >>> However, the release tarball doesn't work because of this bug: >>> https://sourceforge.net/tracker/?func=detail&atid=743020&aid=1518732&group_… >>> <https://sourceforge.net/tracker/?func=detail&atid=743020&aid=1518732&group_…> >>> >>> >>> So, 1. is the cvs version of rel_1_1_0 considered safe for >>> production use? >> >> yes, it is the most recommended version to use -- several issues were >> found and fixed since 1.1.0 release. >> >>> 2. if so and there are important fixes such as this in CVS, >>> would openser.org <http://openser.org> consider >>> releasing a point release containing such changes? >> It is an option, and perhaps should be taken in consideration, as >> patch release with not so much packaging. >> >> This may help some which want to maintain platforms updated, it is >> exampled with 1.0.x, bust just replace the version with 1.1.0 in the >> document and should work (rel_1_0_0 => rel_1_1_0) >> http://openser.org/dokuwiki/doku.php/install:openser-from-cvs >> >> Cheers, >> Daniel >> >> >>> Thanks, >>> >>> Mark Price don lin <don12lin(a)yahoo.com> wrote: Hi there, We have a Postgres Database to store the users' informations before we use Openser. To integrate between Postgres and Openser, we write external C programs and call them from Openser with exec_dset function. The issue we have now is with the Caller ID. When the user makes a call to PSTN number from SIP devices (soft phone or hard phone), the telco carrier requests us to send a numerical Caller ID, not the SIP format (such as john.dow(a)abc.com). What I am doing is to add the Caller ID field into the Postgres for each user and write a C program to query the Caller ID field, then Openser calls it with exec_dset functioin. However, I did not find a way to define a variable in Openser to store this Caller ID and pass it to uac_replace_from function. Appreciate for any suggestions. Jeff, You mentioned that you replaced the From Header, may I ask how you do it ? Thanks and Regards, Don ------------------------------ <Message: 4 <Date: Tue, 19 Dec 2006 11:11:09 +0900 <From: Jeff Williams <jeffw(a)globaldial.com> <Subject: [Users] (PR)ACK problem with uac_replace_from <To: users(a)openser.org <Message-ID: <45874A3D.5070206(a)globaldial.com> <Content-Type: text/plain; charset="iso-8859-1" <I seem to have an issue with uac_replace_from. <I am using openser as a media proxy. For invites I replace the numbers <with leading 0's with 61 for Australia and forward the call to our LCR <box which then forwards the call to the correct voice to pstn gateway. <I am trying to use uac_replace_from to set the from address so the <callerid appears correct on the outgoing calls. <This seems to work fine for the INVITE request, but on subsequent PRACKs <and ACKs (which just get loose routed), something strange happens to <the from address. <On invite, the From get re-written: <From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1790577941 <to <From: Jeff <sip:0892209080@sipone.com>;tag=1790577941 <For the PRACKs and ACKs, this happens: <From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1790577941 <to <From: Jeff <<\032\317\352S\232\222B\230\234K\231*BV\300\030\303\332\ah\303\306i\365\005\016\200\250\000\000\230\016\200\252j\200\002`\000\000\016>;tag=1790577 <941 <There shouldn't be anything happening to the From address on the <(PR)ACKs at all. <Anyone have any ideas? <I have attached a ethereal dump of the call setup. <Jeff __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Users] uri, myself, etc., variables, + is_domain_local()
by Mark Price 22 Dec '06

22 Dec '06

Hi, I've got a few questions: 1. where can I find out what all the variables like uri, myself, etc., mean? 2. how come is_domain_local($ruri) has a different result than uri==myself? In particular, I have my external IP address defined as an alias. I.e.: alias=4.2.2.2 When I say if( ! is_domain_local($ruri) ) when $ruri = user(a)4.2.2.2, the check succeeds. When I say if( ! uri==myself ), the check fails (correctly, I think). Thanks, Mark Price

2 1

[Serusers] mysql tables help
by Katty Xiong 22 Dec '06

22 Dec '06

I am tring to use Ottendorf. I have db tables for 0.9.6 on the machine, so I decide to drop the old tables and create new tables using ser_mysql.sh for Ottendorf, but the script keeps on giving me errors. Do I have to delete the entire old version of SER (0.9.6)? thanks, Joy __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[Users] help needed on avpops
by Zoran Milic 22 Dec '06

22 Dec '06

Hi guys, I am trying to set up radius and avp for ages now, and this time i think i am close to it. I have a problem and I can't find any information about it. The thing is, my soft agent (set to uid 101) can't register, and it is getting "Forbidden IP" even with this in the cfg file (see below) I have set up freeradius and radiusclient-ng according to the manual on http://www.openser.org/docs/openser-radius-1.0.x.html. I can't find anything on the error i am getting in my log. If you need my complete ..cfg or log file, just ask, and I will send them promptly. thanks in advance, best regards, Zoran openser.cfg: .... xlog("L_ERR","rpid = $avp(s:rpid)"); xlog("L_ERR","adresa = $avp(s:adresa)"); xlog("L_ERR","adresa2 = $avp(s:adresa2)"); # check the src ip address # avp_load_radius("caller"); if(!avp_check("$adresa", "eq/$adresa/ig")) { sl_send_reply("403", "Forbidden IP"); exit; }; save("location"); exit; }; .... freeradius/users: .... 101 Auth-Type := Digest, User-Password == "101" Reply-Message = "Authenticated", Sip-Avp += "rpid:101", Sip-Avp += "adresa:10.0.0.157", Sip-Avp += "adresa2:10.0.0.157" .... last entries in the log after the rejected register. .... DEBUG:auth_radius:extract_avp: AVP val is <10.0.0.157> DEBUG:auth_radius:generate_avps: AVP 'adresa'/0='10.0.0.157'/0 has been added DEBUG:auth_radius:extract_avp: string is <adresa2:10.0.0.157> DEBUG:auth_radius:extract_avp: AVP name is <adresa2> DEBUG:auth_radius:extract_avp: AVP val is <10.0.0.157> DEBUG:auth_radius:generate_avps: AVP 'adresa2'/0='10.0.0.157'/0 has been added DEBUG:auth_radius:extract_avp: string is <rpid:101> DEBUG:auth_radius:extract_avp: AVP name is <rpid> DEBUG:auth_radius:extract_avp: AVP val is <101> DEBUG:auth_radius:generate_avps: AVP 'rpid'/0='101'/0 has been added DEBUG:auth_radius:extract_avp: string is <adresa:10.0.0.157> DEBUG:auth_radius:extract_avp: AVP name is <adresa> DEBUG:auth_radius:extract_avp: AVP val is <10.0.0.157> DEBUG:auth_radius:generate_avps: AVP 'adresa'/0='10.0.0.157'/0 has been added DEBUG:auth_radius:extract_avp: string is <adresa2:10.0.0.157> DEBUG:auth_radius:extract_avp: AVP name is <adresa2> DEBUG:auth_radius:extract_avp: AVP val is <10.0.0.157> DEBUG:auth_radius:generate_avps: AVP 'adresa2'/0='10.0.0.157'/0 has been added rpid = 101 adresa = 10.0.0.157 adresa2 = 10.0.0.157 xl_get_spec_value: error - null sp->itf ERROR:avpops:ops_check_avp: cannot get src value parse_headers: flags=ffffffffffffffff check_via_address(192.168.52.145, 192.168.52.145, 0) DEBUG:destroy_avp_list: destroying list 0x404d9238 receive_msg: cleaning up

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Re: [Users] OpenSER as load balancer for several Asterisk servers
by Edoardo Serra 22 Dec '06

22 Dec '06

Hi Users, I'm here again. After having read more docs I don't feel like using rtpproxy My Asterisks have both public IP Adresses so I don't need to proxy rtp stream I just need a configuration to randomly proxy SIP requests to Asterisk boxes behind SER (for load balancing purposes) and to handle registrations of user to forward incoming calls to them (calls are coming from Asterisks to OpenSER) I'm really stuck on it because I already have such a working setup, with same config files. I'm just trying to replicate it on another site Tnx in advance for help Edoardo My last mail to the ml follow for reference Tnx again Klaus, i'll read docs more in depth and try to understand each change to openser.cfg you suggested >>U 2006/12/20 19:15:35.025446 OOO.OOO.OOO.OOO:5060 -> CCC.CCC.CCC.CCC:21722 >>SIP/2.0 183 Session Progress. >>Via: SIP/2.0/UDP >>OOO.OOO.OOO.OOO:5060;branch=z9hG4bK-d87543-3e229802603d7c32-1--d87543-;rport=21722. >> > >^^^^^^^^^^^^^^ > >strange bug again - there must be CCC.CCC.CCC.CCC really strange, do you think it's an openser bug ? The strangest thing is that I just copied openser configs from a working system (Openser + Asterisks) changing just ip addresses OpenSER version is also the same... [...] >># !! Nathelper >>onreply_route[1] { >> # NATed transaction ? >> if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { >> fix_nated_contact(); >> #force_rtp_proxy(); > ^ >Are you sure it is commented? I do not believe it because the ngrep >shows that the SDP of 200 Ok is rewritten. Sure, I also tried to remove the line without solving the previous problem >Please read the Getting Started Turial from onsip.org carefully. It >describes how you handle NAT correctly and also describes how to do >NAT traversal for in-dialog messages, which is missing in your config. I will. Tnx again Regards Edoardo

1 1

Re: [Users] Load-Balancing
by raviprakash sunkara 22 Dec '06

22 Dec '06

May i guess U Perform the Dispatcher modules ..... May this Hopes u to get On 12/22/06, raviprakash sunkara <sunkara.raviprakash.feb14(a)gmail.com> wrote: > > > > > > On 12/22/06, Thomas Deillon < Thomas.Deillon(a)smart-telecom.ch> wrote: > > > > Hi, > > > > I'm trying to set a load balancing solution with SER and twos Asterisk. > > The Asterisks will be use as REGISTRAR, REDIRECT, GATEWAY .... > > > > The ser will just load balance the traffic on one Asterisk or an other. > > I use the dispatcher module for the SIP message and I use 2 public ip > > addresses for the asterisk and so, I'm able to directly send the RTP > > traffic to them (I didn't use mediaproxy or rtpproxy). > > > > On each asterisk, I set the SER as default GW to be sure that all > > message go through this computer. (Client are behind a FW/NAT and so, I > > would like to be sure that all message come from the same IP > > address:port else, the FW/NAT will DROP these messages). > > > > For the moment, with this solution, I have a problem. The message from > > client will go to SER which will resend it to the asterisk and then, the > > > > asterisk will try to answer responding on the same IP -> Loop detected > > !! > > > > On some device I saw that we can forward the request for the phone to > > the asterisk without changing the source ip address .... > > > > Do you know how can I do this ? > > > > Thanks a lot for your help, > > > > Thomas D > > > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > -- > Thanks and Regards > Ravi Prakash Sunkara > ravi.sunkara(a)hyperion-tech.com > M:+91 9985077535 > O:+91 40 23114549 > F:+91 40 40208727 > ravi.sunkara(a)hyperion-tech.com > www.hyperion-tech.com -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

1 0

[Users] Routing calls to a sip terminator
by Xantek, Inc. 22 Dec '06

22 Dec '06

I am sure this must have been covered somewhere, but I cannot find it. I am trying to get openser to register with username and password and forward a call to that server. This is for outbound sip termination to the PSTN. If somebody could point me to the right place, I would be grateful. Bill

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[Users] Load-Balancing
by Thomas Deillon 21 Dec '06

21 Dec '06

Hi, I'm trying to set a load balancing solution with SER and twos Asterisk. The Asterisks will be use as REGISTRAR, REDIRECT, GATEWAY .... The ser will just load balance the traffic on one Asterisk or an other. I use the dispatcher module for the SIP message and I use 2 public ip addresses for the asterisk and so, I'm able to directly send the RTP traffic to them (I didn't use mediaproxy or rtpproxy). On each asterisk, I set the SER as default GW to be sure that all message go through this computer. (Client are behind a FW/NAT and so, I would like to be sure that all message come from the same IP address:port else, the FW/NAT will DROP these messages). For the moment, with this solution, I have a problem. The message from client will go to SER which will resend it to the asterisk and then, the asterisk will try to answer responding on the same IP -> Loop detected !! On some device I saw that we can forward the request for the phone to the asterisk without changing the source ip address .... Do you know how can I do this ? Thanks a lot for your help, Thomas D

1 0

[Serusers] Re: How to change Caller ID in FROM field ?
by lawrence k.y. lin 21 Dec '06

21 Dec '06

Hi all, Appreciate if someone can give me a suggestion or even an answer. Since the call from SIP device to PSTN phone did not have Caller ID and our Telco carriers request for the Caller ID, we have to set a number before the calls are routed to PSTN gateway. I did it in Asterisk before with the function calls SetCallerID(CLID) or Set(CALLERID(number)=CLID) (depends on the Asterisk version) and these function calls worked. But because of the scalabliity issue, I changed to SER and would like to do the similar thing as in Asterisk. I read the SER document and awared SIP_HF_FROM has the data for the entired FROM field. Is there any other variables for Caller ID only ? I searched the mailing list for SER and did not find any related topic (only found one with title: caller-id with raius using sip-rpid, but it is irrelevant). Thanks in advance. Larry _________________________________________________________________ Try the next generation of search with Windows Live Search today! http://imagine-windowslive.com/minisites/searchlaunch/?locale=en-us&source=…

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[Users] Realtime Call Monitoring
by lists＠infoway.net 21 Dec '06

21 Dec '06

We are Asterisk-only users and currently use Asterisk Flash Operator Panel (http://www.asternic.org/) to get a real-time view of the different phones/extensions, channels, etc. We are planning on enhancing our setup by adding SER/OpenSER in front of our Asterisk systems and to expand into other venues as well. If we wanted to use something like SEMS instead of Asterisk for certain services, is there any tool similar to the Flash Operator Panel (FOP) available for SER/OpenSER + SEMS? I guess we could always continue using FOP with our Asterisk systems fronted by SER/OpenSER. However, for our other plans where we prefer to use SEMS, we would love to see if there is any solution available (free, sponsored, or commercial). Thanks, Daniel

4 6

[Users] Re: any kind of statistics available
by Daniel-Constantin Mierla 21 Dec '06

21 Dec '06

Just to complete the thread, a new SNMP module is available for such purpose. In the announcement are listed several tools that can help to collect statistics and generate grphs. It is a matter of configuration and maybe small shell scripting: http://openser.org/pipermail/users/2006-December/008334.html http://cacti.net/ http://oss.oetiker.ch/mrtg/ At the first sight these tools can be used with FIFO or XMLRPC interface as well, if you write the shell script to collect the statistics. Cheers, Daniel Wienand, Joerg wrote: >> You can make a small script to pull periodically statistics via >> FiFO/XMLRPC into a database/file and from there generate graphics at >> your convenience. >> > > This is what I thought about. I expect that I'm not the only one who is > interested in statistics. So maybe anyone else has done this job before. ;-) > > regards, Jörg > >

1 0

Re: [Users] Problem with avp_check
by Bogdan-Andrei Iancu 21 Dec '06

21 Dec '06

Hi Jose, I think your regexp is not correct - you need to force the end of string, otherwise only a substring (prefix) will be matched. try avp_check("$fU","re/^[0-9]{10}$/g") regards, bogdan Jose Gil Navarrete wrote: > Hi thanks Andrei, I tried with your recomendation but I obtain the > same result, if $fu is: 012345678901, then "To-username > -012345678901-" is inserted > > if(avp_check("$fU","re/^[0-9]{10}/g")) { > insert_hf("To-username: -$rU- \r\n", "Call-ID"); > } else { > sl_send_reply("400", "Peticion Erronea, numero > invalido"); > }; > > > */Bogdan-Andrei Iancu <bogdan(a)voice-system.ro>/* escribió: > > Hi Jose, > > the FM operator in avp_check is not for regular expressions - FM is a > file matching- like operator (like "sip:*@domain). See "man 3 > fnmatch". > > if you want to check against an regexp, you need to use the RE > operator. > See: > http://www.openser.org/docs/modules/1.2.x/avpops.html#AEN384 > > as RE op does not support an AVP as right operand, you should try: > if(avp_check("$fU","re/^[0-9]{10}/g")) { > > regards, > bogdan > > Jose Gil Navarrete wrote: > > > Hi > > I try to check the username in the URI, this is, if the username do > > not have ten digits then send a message, I made: > > > > avp_write("^[0-9]{10}","$avp(s:fm_avp)"); > > if(avp_check("$fU","fm/$avp(s:fm_avp)/g")) { > > insert_hf("To-username: -$fU- \r\n", "Call-ID"); > > } else { > > sl_send_reply("400", "Bad Request"); > > exit; > > }; > > > > but I ever obtain 400 message. > > > > I try too with: > > if(avp_check("$fU","re/[0-9]{10}$/g")) { > > insert_hf("To-username: -$fU- \r\n", "Call-ID"); > > else{ > > sl_send_reply("400", "Bad request"); > > exit; > > } > > > > But I obtain the same message. Any idea? > > > > __________________________________________________ > > Correo Yahoo! > > Espacio para todos tus mensajes, antivirus y antispam ¡gratis! > > Regístrate ya - http://correo.yahoo.com.mx/ > > > >------------------------------------------------------------------------ > > > >_______________________________________________ > >Users mailing list > >Users(a)openser.org > >http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > __________________________________________________ > Correo Yahoo! > Espacio para todos tus mensajes, antivirus y antispam ¡gratis! > Regístrate ya - http://correo.yahoo.com.mx/ >

1 0

[Serusers] SER 0.9.x and 0.10.x on the same machine?
by TZieleniewski 21 Dec '06

21 Dec '06

Hi ALL!! Is it possible to have to different releases of ser on the same machine? Have any one tired that? I couldn't find any info the web, please help :) Thanks Bests tomek

3 3

[Users] OpenSER as load balancer for several Asterisk servers
by Edoardo Serra 21 Dec '06

21 Dec '06

Hi guys, I'm having a problem with an OpenSER acting as registrar server and load balancer for many Asterisk servers. In a few words: "users are registering on openser and, when they want to make a call, OpenSER proxies the request to an Asterisk server with the dispatcher module" Here is the intended data flow (SIP goes through OpenSER and media goes directly to Asterisk) User <-- SIP --> OpenSER <-- SIP --> Asterisk User <-- RTP --> Asterisk Both, OpenSER and Asterisks have public IPs I already have a working setup of that and everything seems working correctly. I'm trying to replicate that setup on another site, same configurations of the boxes, same versions of OpenSER and Asterisk, etc... but I'm having monodirectional Audio. Having a look with tethereal I see that OpenSER, when the communication is answered, sends a SIP packet (200 OK) to the user indicating itself as media endpoint instead of the Asterisks. From that moment I see RTP packets flowing from the client to OpenSER This seems really strange to me because I just copied the same configurations file from a working setup to the new installation. Tnx in advance for help. Regards P.S.: Here is my openser.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=3 # debug level (cmd line: -dddddddddd) fork=yes #log_stderror=no # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #children=4 #port=5060 fifo="/tmp/ser_fifo" #uid=nobody #gid=nobody # ------------------ module loading ---------------------------------- loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/usrloc.so" loadmodule "/usr/lib/openser/modules/registrar.so" loadmodule "/usr/lib/openser/modules/nathelper.so" loadmodule "/usr/lib/openser/modules/textops.so" loadmodule "/usr/lib/openser/modules/exec.so" loadmodule "/usr/lib/openser/modules/uri.so" loadmodule "/usr/lib/openser/modules/uri_db.so" loadmodule "/usr/lib/openser/modules/dispatcher.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/openser/modules/mysql.so" loadmodule "/usr/lib/openser/modules/auth.so" loadmodule "/usr/lib/openser/modules/auth_db.so" modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://xxx:xxx@xxx.xxx.xxx.xxx/openser") modparam("usrloc", "timer_interval", 120) modparam("auth_db", "calculate_ha1", 0) modparam("auth_db", "db_url", "mysql://xxx:xxx@xxx.xxx.xxx.xxx/voip") modparam("uri_db", "db_url", "mysql://xxx:xxx@xxx.xxx.xxx.xxx/openser") modparam("rr", "enable_full_lr", 1) modparam("registrar", "nat_flag", 6) modparam("registrar", "max_expires", 3600) modparam("registrar", "min_expires", 60) modparam("registrar", "append_branches", 0) modparam("registrar", "desc_time_order", 1) modparam("nathelper", "natping_interval", 20) # Ping interval 20 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT modparam("dispatcher", "force_dst", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); exit; }; if ( (method=="OPTIONS") || (method=="SUBSCRIBE") || (method=="NOTIFY") ) { sl_send_reply("405", "Method Not Allowed"); exit; } if (!method=="REGISTER") { record_route(); }; if ((src_ip==xxx.xxx.xxx.xxx) || (src_ip==xxx.xxx.xxx.xxx)) { # IP of Asterisks if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; # 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(); }; exit; }; if (nat_uac_test("3")) { if ((method=="REGISTER") || (method=="INVITE") || (method=="OPTIONS")) { fix_nated_contact(); force_rport(); setflag(6); # Mark as NATed } } # 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 (method=="REGISTER") { if (!proxy_authorize("domain", "openser_view")) { proxy_challenge("domain", "0"); exit; } if (!check_to()) { sl_send_reply("403", "Digest username and URI username do NOT match! Stay away!"); exit; } save("location"); exit; }; if (method=="INVITE") { if (!proxy_authorize("domain", "openser_view")) { proxy_challenge("domain", "0"); exit; } if (!check_from()) { sl_send_reply("403", "Digest username and URI username do NOT match! Stay away!"); exit; } } # loose-route processing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); exit; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); exit; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # ! Nathelper if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); exit; }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if ((src_ip!=xxx.xxx.xxx.xxx) && (src_ip!=xxx.xxx.xxx.xxx)) { # IP of Asterisks ds_select_dst("2", "0"); } if (!t_relay()) { sl_reply_error(); }; } # ! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); }; }

4 13

[Serusers] Multiple domain
by TZieleniewski 21 Dec '06

21 Dec '06

Hi ! A have a small technical question:) It considers serving of multiple domain. I would like to make sure that I understand things right. After I read the user doc and README from domain module I came to the following state of art:) 1)one has to create the seperate location and subscriber table for each domain *so if i have for instance two domains : dom1.com and dom2.com then my ser has to have the according dns records in the dns server? 2) after creating the tables i need to add domains record in the database to the domain table *I dont know why but when ever i try to add or show domains o get the following error 500 command 'domain_dump' not available - why is that so? any way instead of an error the serctl command adds the domain record to the database 3) modify the ser.cfg file - add appropriate modparam commands 4) modify the ser.cfg file by adding a regular expression analizys to invoke the save, www_chalenge, proxy_chalenge and lookup correctly with the right domain? Please tell me is my thinking here right? or maybe there is another way As allway thank u for your time. Bests tomek

3 3

Re: [Users] Delayed signalling?
by Jiri Kuthan 21 Dec '06

21 Dec '06

if you happen to have a PCAP file with the incident, let me please know. -jiri p.s. even if you didn't tweak timers, the results may be suboptimal because the software version you are using is having rather indeterministic timer subsystem. For example, the recent measurements (http://www.iptel.org/ser/doc/performance) show quite scattered server responsiveness under high load. (Note though that the measurement results were achieved in a best-effort manner based on the tester knowledge and understanding of openser and that the result are not officially confirmed by the OpenSER project.) Whether it is indeed the cause is not certain though -- this looks really like a stealth bug. At 15:58 15/12/2006, Max Gregorian wrote: >Thanks very much for all the replies. I shall try and post a config and traces as soon as I can get them from the office. > >Some more information, if it helps: > >Server specs: >- HP ProLiant DL360 G4 (1U rack servers) >- 3 GHz processors (800 MHz FSB) >- 1 GB RAM >- 10K rpm SCSI HDs (in a RAID 1+0 Mirror) > ># Servers are running OpenSER 1.0.1 (no-TLS). ># Servers are listening on 3 ports (both tcp and udp for each port), so in openserctl ps I am seeing 4 child processes for each port. ># Servers running CentOS Linux 4.3 ># MySQL installed when CentOS was installed but not running and not currently being used with Openser. > > >Things I have pretty much managed to eliminate are: >1. It's doesn't seem to be hardware. The specs for the servers are more than sufficient I think. >2. It doesn't seem to be traffic/load related as I see these problems on 2 brand new servers I have just installed with no traffic on them. However, it does seem to get worse with more traffic. >3. I don't think it's database related as I have deliberately not configured mysql on any of the servers in case of database performance. >4. I haven't played with the timers at all so far. >5. I haven't configured nscd yet, but as far as I can tell it's not caching DNS. >6. Though openser is listening on tcp ports as well, currently only the udp ports are being used as most of our customers use hardware phones. In any case, I haven't as yet seen as requests on tcp. >7. I am not sure it is DNS as in the tests I ran I sent requests directly to the external IP of the server and not to the domain name it is responsible for. Also the test servers are now only responsible for one domain, but in future will have more than one. >8. Also TTL on the domain name is really short. Ping from the server itself TTL=64 and ping times are low as you would expect (< 1ms when pinging from the server itself). Ping from outside the network (from the internet - for me - tp the domain was) 12ms (average), no packet loss, TTL = 53. >9. I have not setup any internal DNS entries for the domain. Servers are resolving domain from entries in /etc/hosts. > >Like I said, it doesn't happen all the time - just maybe once or twice every hour on the servers with more traffic. > >I ran SIPp pointing at one of the new servers last week and at around 100CPS I was seeing about 2,000 out of approx. 10,000 calls were failing. Setup was UAC -> openser -> UAS (Both UAC and UAS were running on the same machine, but different ports). Again there is no traffic on these servers now so I have no idea why so many failed calls. > >I am not sure if any of this information helps, but I am certainly open to suggestions on things to try. > >Thanks in advance. > > > >On 12/14/06, samuel <<mailto:samu60@gmail.com>samu60(a)gmail.com> wrote: >It might be due to a DNS query....whenver a request has to be >forwarded to a domain, openSER makes a DNS query to resolv the IP. >During this operation, the child processing the request will not >answer to further incoming messages. > >it also can be happening due to a spiral loop that stays on the server. > >Without further information (confg,logs) it's hard to tell which is >the reason... > >hope it helps, >Samuel. > >without more information > >2006/12/14, Max Gregorian <<mailto:gregorian442@googlemail.com>gregorian442(a)googlemail.com>: >> Hi all, >> >> Just wondering if anyone else has had this problem. I have noticed while >> tracing on my OpenSER server, that every now and then the server receives a >> packet which it does to respond to immediately, resulting in a string of >> packets being sent to the server and then the server responding a few >> seconds later. This does not happen all the time, just say maybe once or >> twice every hour. The rest of the time the signaling is correct and >> responses follow request packets in the correct order. >> >> What I am trying to figure out is whether this is a load traffic issue (i.e. >> can the server not handle too much load), and if so is it OpenSER or the >> network or the server in general? I have run diagnostics on the servers and >> there is nothing wrong with the hardware. >> >> On the other hand Could this be related to any timer issues? I remember >> there was mention of timers in SER but are there any default timer settings >> that can be tweaked? >> >> Thanks in advance for any response. >> >> >> _______________________________________________ >> Users mailing list >> <mailto:Users@openser.org>Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> > > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users -- Jiri Kuthan http://iptel.org/~jiri/

4 4

[Users] Problem with avp_check
by Jose Gil Navarrete 21 Dec '06

21 Dec '06

Hi I try to check the username in the URI, this is, if the username do not have ten digits then send a message, I made: avp_write("^[0-9]{10}","$avp(s:fm_avp)"); if(avp_check("$fU","fm/$avp(s:fm_avp)/g")) { insert_hf("To-username: -$fU- \r\n", "Call-ID"); } else { sl_send_reply("400", "Bad Request"); exit; }; but I ever obtain 400 message. I try too with: if(avp_check("$fU","re/[0-9]{10}$/g")) { insert_hf("To-username: -$fU- \r\n", "Call-ID"); else{ sl_send_reply("400", "Bad request"); exit; } But I obtain the same message. Any idea? __________________________________________________ Correo Yahoo! Espacio para todos tus mensajes, antivirus y antispam ¡gratis! Regístrate ya - http://correo.yahoo.com.mx/

2 1

[Users] flags in reply route
by Klaus Darilion 21 Dec '06

21 Dec '06

Hi! Do I have access to the flags of a transaction in a reply route? regards klaus -- Klaus Darilion nic.at

2 1

Re: [Serusers] branch parameter
by Tuffy Dhawan 21 Dec '06

21 Dec '06

SER is identifying call even W/O branch parameter and i have not used forward(). My only question was to make sure that RTC/1.2 should generate branch parameter. >From: Martin Hoffmann <hn(a)nvnc.de> >To: Jiri Kuthan <jiri(a)iptel.org> >CC: Kapil Dhawan <digitaltuffy(a)hotmail.com>, serusers(a)lists.iptel.org >Subject: Re: [Serusers] branch parameter >Date: Mon, 18 Dec 2006 16:58:00 +0100 > >Jiri Kuthan wrote: > > At 17:05 17/12/2006, Kapil Dhawan wrote: > > > > > I am using RTC/1.2 library and its not generating branch parameter. > > > > too bad, most implementation can deal with it in backw2ards >compatibility mode > > but clients should no longer generate such branch. I would be looking >for > > a more up-to-date stack if I was you. > ><evil> >Ever used forward() in SER? ></evil> > >Regards, >Martin _________________________________________________________________ Christmas is drawing near and the excitement is in the air! Check out this cool photofeature on Christmas celebrations the world over http://www.msnspecials.in/photogallery/photogallery.asp?id=a200611814137906

4 4

Re: [Users] How to get CDR in Openser
by Klaus Darilion 21 Dec '06

21 Dec '06

Howard Tang wrote: > Hi klaus, > > I am new to the VOIP / Voice world. Thank you for your reply. I just > want openser to generate CDR. Do i need radius to do that? > > Regards, > Howard > > Klaus Darilion wrote: >> Howard Tang wrote: >>> Hi, >>> >>> Can someone give me some idea of how i can obtain CDR from Openser >>> please? >>> >>> I saw a open source software called cdrtools, however, i don't want >>> to install apache on my machine. >> >> cdrtools does not generate CDRs. It just analyzes the CDRs. >> >> Thus, what do you want? >> Creating CDRs or analyzing? >> >> openser can create CDRs for DB, Radius or syslog using the acc module >> (see module documentation on openser.org) no, you do not need radius. you can alos use syslog accounting or database accounting. http://www.openser.org/docs/modules/1.1.x/acc.html btw: please cc the mailing list regards klaus >> >> regards >> klaus >> >>> >>> Any other CDR software will work with openser >> >>> >>> Regards, >>> Howard >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)openser.org >>> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > -- Klaus Darilion nic.at

2 2

[Users] A question about cancels..
by nick 21 Dec '06

21 Dec '06

When I make a call to someone, a call which is correctly connected, for example via PSTN gateway, I can watch INVITEs, ACKs, OKs and BYEs being relayed from my station the my openser to the gateway, and back.. However if the phone rings, and I decide to terminate the call (before the other person responds), I see the CANCEL going from my station to the sipserver, but I don't see it being sent to the PSTN gateway.. Is this something I have to specifically route for? The CANCEL is part of the same call sequence, is there a reason it wouldn't be relayed, other than a mistake in my routing, obviously.. Nick

2 1

Re: [Serusers] Problem with MTU size in SER with MIPv6.
by Jose Angel Calvo 21 Dec '06

21 Dec '06

Now, i'll explain and show the INVITE causing the crash: The INVITE packet is fragmented in two frames. The capture is done with the ethereal (wireshark now) and it's the next: *********************************************************************** No. Time Source Destination Protocol Info 1 0.000000 2001:720:1500:24:fcfd:ff:fe0d:101 2001:720:1500:1b:20c:76ff:fe6c:4658 IPv6 IPv6 fragment (nxt=UDP (0x11) off=1408 id=0x5) Frame 1 (431 bytes on wire, 431 bytes captured) Arrival Time: Nov 30, 2006 16:40:02.345512000 Time delta from previous packet: 0.000000000 seconds Time since reference or first frame: 0.000000000 seconds Frame Number: 1 Packet Length: 431 bytes Capture Length: 431 bytes Protocols in frame: sll:ipv6:data Linux cooked capture Packet type: Unicast to another host (3) Link-layer address type: 1 Link-layer address length: 6 Source: Cisco_17:28:70 (00:d0:06:17:28:70) Protocol: IPv6 (0x86dd) Internet Protocol Version 6 Version: 6 Traffic class: 0x00 Flowlabel: 0x00000 Payload length: 375 Next header: IPv6 fragment (0x2c) Hop limit: 61 Source address: 2001:720:1500:24:fcfd:ff:fe0d:101 (2001:720:1500:24:fcfd:ff:fe0d:101) Destination address: 2001:720:1500:1b:20c:76ff:fe6c:4658 (2001:720:1500:1b:20c:76ff:fe6c:4658) Fragmentation Header Next header: UDP (0x11) Offset: 1408 More fragments: No Identification: 0x00000005 Data (367 bytes) 0000 00 03 00 01 00 06 00 d0 06 17 28 70 00 00 86 dd ..........(p.... 0010 60 00 00 00 01 77 2c 3d 20 01 07 20 15 00 00 24 `....w,= .. ...$ 0020 fc fd 00 ff fe 0d 01 01 20 01 07 20 15 00 00 1b ........ .. .... 0030 02 0c 76 ff fe 6c 46 58 11 00 05 80 00 00 00 05 ..v..lFX........ 0040 50 44 41 54 45 0d 0a 52 65 70 6c 61 63 65 73 3a PDATE..Replaces: 0050 20 64 34 63 38 37 63 39 30 32 64 31 62 35 32 62 d4c87c902d1b52b 0060 32 38 36 62 61 64 39 36 66 33 63 62 62 36 31 34 286bad96f3cbb614 0070 34 40 5b 32 30 30 31 3a 37 32 30 3a 31 35 30 30 4@[2001:720:1500 0080 3a 31 61 3a 32 33 30 3a 31 62 66 66 3a 66 65 31 :1a:230:1bff:fe1 0090 30 3a 35 66 62 33 5d 3b 72 65 61 73 6f 6e 3d 53 0:5fb3];reason=S 00a0 49 50 20 72 6f 75 74 65 20 68 61 73 20 63 68 61 IP route has cha 00b0 6e 67 65 64 0d 0a 43 6f 6e 74 65 6e 74 2d 54 79 nged..Content-Ty 00c0 70 65 3a 20 61 70 70 6c 69 63 61 74 69 6f 6e 2f pe: application/ 00d0 73 64 70 0d 0a 43 6f 6e 74 65 6e 74 2d 4c 65 6e sdp..Content-Len 00e0 67 74 68 3a 20 31 39 35 0d 0a 0d 0a 76 3d 30 0d gth: 195....v=0. 00f0 0a 6f 3d 69 73 61 62 65 6c 20 30 20 30 20 49 4e .o=isabel 0 0 IN 0100 20 49 50 36 20 5b 32 30 30 31 3a 37 32 30 3a 31 IP6 [2001:720:1 0110 35 30 30 3a 31 61 3a 32 33 30 3a 31 62 66 66 3a 500:1a:230:1bff: 0120 66 65 31 30 3a 35 66 62 33 5d 0d 0a 73 3d 2d 0d fe10:5fb3]..s=-. 0130 0a 63 3d 49 4e 20 49 50 36 20 32 30 30 31 3a 37 .c=IN IP6 2001:7 0140 32 30 3a 31 35 30 30 3a 31 61 3a 32 33 30 3a 31 20:1500:1a:230:1 0150 62 66 66 3a 66 65 31 30 3a 35 66 62 33 0d 0a 74 bff:fe10:5fb3..t 0160 3d 30 20 30 0d 0a 6d 3d 61 75 64 69 6f 20 32 32 =0 0..m=audio 22 0170 32 32 34 20 52 54 50 2f 41 56 50 20 30 20 33 20 224 RTP/AVP 0 3 0180 34 20 35 20 36 20 38 20 31 35 20 31 38 0d 0a 6d 4 5 6 8 15 18..m 0190 3d 76 69 64 65 6f 20 32 32 32 32 32 20 52 54 50 =video 22222 RTP 01a0 2f 41 56 50 20 32 36 20 33 34 20 33 31 0d 0a /AVP 26 34 31.. No. Time Source Destination Protocol Info 2 0.000117 2001:720:1500:24:fcfd:ff:fe0d:101 2001:720:1500:1b:20c:76ff:fe6c:4658 SIP Request: INVITE sip:[2001:720:1500:1b:20c:76ff:fe6c:4658]:5060;transport=udp Frame 2 (1512 bytes on wire, 1512 bytes captured) Arrival Time: Nov 30, 2006 16:40:02.345629000 Time delta from previous packet: 0.000117000 seconds Time since reference or first frame: 0.000117000 seconds Frame Number: 2 Packet Length: 1512 bytes Capture Length: 1512 bytes Protocols in frame: sll:ipv6:ipv6:udp:sip Linux cooked capture Packet type: Sent by us (4) Link-layer address type: 1 Link-layer address length: 6 Source: fe:fd:00:0c:00:01 (fe:fd:00:0c:00:01) Protocol: IPv6 (0x86dd) Internet Protocol Version 6 Version: 6 Traffic class: 0x00 Flowlabel: 0x00000 Payload length: 1456 Next header: IPv6 (0x29) Hop limit: 64 Source address: 2001:720:1500:1b:fcfd:ff:fe0c:1 (2001:720:1500:1b:fcfd:ff:fe0c:1) Destination address: 2001:720:1500:24:20c:76ff:fe6c:4658 (2001:720:1500:24:20c:76ff:fe6c:4658) Internet Protocol Version 6 Version: 6 Traffic class: 0x00 Flowlabel: 0x00000 Payload length: 1416 Next header: IPv6 fragment (0x2c) Hop limit: 60 Source address: 2001:720:1500:24:fcfd:ff:fe0d:101 (2001:720:1500:24:fcfd:ff:fe0d:101) Destination address: 2001:720:1500:1b:20c:76ff:fe6c:4658 (2001:720:1500:1b:20c:76ff:fe6c:4658) Fragmentation Header Next header: UDP (0x11) Offset: 0 More fragments: Yes Identification: 0x00000005 User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Source port: sip (5060) Destination port: sip (5060) Length: 1775 Checksum: 0x9301 Session Initiation Protocol Request-Line: INVITE sip:[2001:720:1500:1b:20c:76ff:fe6c:4658]:5060;transport=udp SIP/2.0 Method: INVITE Resent Packet: False Message Header Record-Route: <sip:[2001:720:1500:24:FCFD:FF:FE0D:101];ftag=12122347;lr=on> Record-Route: <sip:[2001:720:1500:1B:FCFD:FF:FE0C:103];ftag=12122347;lr=on> Record-Route: <sip:[2001:720:1500:1A:FCFD:FF:FE0A:103];ftag=12122347;lr=on> Record-Route: <sip:[2001:720:1500:19:FCFD:FF:FE0E:101];ftag=12122347;lr=on> Call-ID: 138104335821ad9b21971c7d6059b6c8@[2001:720:1500:1a:230:1bff:fe10:5fb3] CSeq: 1 INVITE From: "ueA" <sip:ueA@domainA.ammi.dit.upm.es>;tag=12122347 SIP Display info: "ueA" SIP from address: sip:ueA@domainA.ammi.dit.upm.es SIP tag: 12122347 To: <sip:ueC@domainC.ammi.dit.upm.es> SIP to address: sip:ueC@domainC.ammi.dit.upm.es Via: SIP/2.0/UDP [2001:720:1500:24:FCFD:FF:FE0D:101];branch=z9hG4bK1205.ddbf87e4.0 Via: SIP/2.0/UDP [2001:720:1500:1B:FCFD:FF:FE0C:103];branch=z9hG4bK1205.c8a66626.0 Via: SIP/2.0/UDP [2001:720:1500:1B:FCFD:FF:FE0C:102];branch=z9hG4bK1205.1cb3a0b4.0 Via: SIP/2.0/UDP [2001:720:1500:1A:FCFD:FF:FE0A:103];branch=z9hG4bK1205.44344f04.0 Via: SIP/2.0/UDP [2001:720:1500:19:FCFD:FF:FE0E:101];branch=z9hG4bK1205.a44e3b34.0 Via: SIP/2.0/UDP [2001:720:1500:1a:230:1bff:fe10:5fb3]:61052;branch=z9hG4bK40e1844512a234e326ef8222f8b204fb Max-Forwards: 12 Contact: "ueA" <sip:[2001:720:1500:1a:230:1bff:fe10:5fb3]:61052;transport=udp> Contact Binding: "ueA" <sip:[2001:720:1500:1a:230:1bff:fe10:5fb3]:61052;transport=udp> URI: "ueA" <sip:[2001:720:1500:1a:230:1bff:fe10:5fb3]:61052;transport=udp> SIP Display info: "ueA" SIP contact address: sip:[2001:720:1500:1a:230:1bff:fe10:5fb3]:61052 P-Preferred-Identity: "ueA" <sip:ueA@domainA.ammi.dit.upm.es>;tag=12122347 Privacy: none P-Access-Network-Info: 3GPP-UTRAN-TDD;utran-cell-id-3gpp=181086 Allow: INVITE,ACK,CANCEL,BYE,PRACK,U 0000 00 04 00 01 00 06 fe fd 00 0c 00 01 00 00 86 dd ................ 0010 60 00 00 00 05 b0 29 40 20 01 07 20 15 00 00 1b `.....)@ .. .... 0020 fc fd 00 ff fe 0c 00 01 20 01 07 20 15 00 00 24 ........ .. ...$ 0030 02 0c 76 ff fe 6c 46 58 60 00 00 00 05 88 2c 3c ..v..lFX`.....,< 0040 20 01 07 20 15 00 00 24 fc fd 00 ff fe 0d 01 01 .. ...$........ 0050 20 01 07 20 15 00 00 1b 02 0c 76 ff fe 6c 46 58 .. ......v..lFX 0060 11 00 00 01 00 00 00 05 13 c4 13 c4 06 ef 93 01 ................ 0070 49 4e 56 49 54 45 20 73 69 70 3a 5b 32 30 30 31 INVITE sip:[2001 0080 3a 37 32 30 3a 31 35 30 30 3a 31 62 3a 32 30 63 :720:1500:1b:20c 0090 3a 37 36 66 66 3a 66 65 36 63 3a 34 36 35 38 5d :76ff:fe6c:4658] 00a0 3a 35 30 36 30 3b 74 72 61 6e 73 70 6f 72 74 3d :5060;transport= 00b0 75 64 70 20 53 49 50 2f 32 2e 30 0d 0a 52 65 63 udp SIP/2.0..Rec 00c0 6f 72 64 2d 52 6f 75 74 65 3a 20 3c 73 69 70 3a ord-Route: <sip: 00d0 5b 32 30 30 31 3a 37 32 30 3a 31 35 30 30 3a 32 [2001:720:1500:2 00e0 34 3a 46 43 46 44 3a 46 46 3a 46 45 30 44 3a 31 4:FCFD:FF:FE0D:1 00f0 30 31 5d 3b 66 74 61 67 3d 31 32 31 32 32 33 34 01];ftag=1212234 0100 37 3b 6c 72 3d 6f 6e 3e 0d 0a 52 65 63 6f 72 64 7;lr=on>..Record 0110 2d 52 6f 75 74 65 3a 20 3c 73 69 70 3a 5b 32 30 -Route: <sip:[20 0120 30 31 3a 37 32 30 3a 31 35 30 30 3a 31 42 3a 46 01:720:1500:1B:F 0130 43 46 44 3a 46 46 3a 46 45 30 43 3a 31 30 33 5d CFD:FF:FE0C:103] 0140 3b 66 74 61 67 3d 31 32 31 32 32 33 34 37 3b 6c ;ftag=12122347;l 0150 72 3d 6f 6e 3e 0d 0a 52 65 63 6f 72 64 2d 52 6f r=on>..Record-Ro 0160 75 74 65 3a 20 3c 73 69 70 3a 5b 32 30 30 31 3a ute: <sip:[2001: 0170 37 32 30 3a 31 35 30 30 3a 31 41 3a 46 43 46 44 720:1500:1A:FCFD 0180 3a 46 46 3a 46 45 30 41 3a 31 30 33 5d 3b 66 74 :FF:FE0A:103];ft 0190 61 67 3d 31 32 31 32 32 33 34 37 3b 6c 72 3d 6f ag=12122347;lr=o 01a0 6e 3e 0d 0a 52 65 63 6f 72 64 2d 52 6f 75 74 65 n>..Record-Route 01b0 3a 20 3c 73 69 70 3a 5b 32 30 30 31 3a 37 32 30 : <sip:[2001:720 01c0 3a 31 35 30 30 3a 31 39 3a 46 43 46 44 3a 46 46 :1500:19:FCFD:FF 01d0 3a 46 45 30 45 3a 31 30 31 5d 3b 66 74 61 67 3d :FE0E:101];ftag= 01e0 31 32 31 32 32 33 34 37 3b 6c 72 3d 6f 6e 3e 0d 12122347;lr=on>. 01f0 0a 43 61 6c 6c 2d 49 44 3a 20 31 33 38 31 30 34 .Call-ID: 138104 0200 33 33 35 38 32 31 61 64 39 62 32 31 39 37 31 63 335821ad9b21971c 0210 37 64 36 30 35 39 62 36 63 38 40 5b 32 30 30 31 7d6059b6c8@[2001 0220 3a 37 32 30 3a 31 35 30 30 3a 31 61 3a 32 33 30 :720:1500:1a:230 0230 3a 31 62 66 66 3a 66 65 31 30 3a 35 66 62 33 5d :1bff:fe10:5fb3] 0240 0d 0a 43 53 65 71 3a 20 31 20 49 4e 56 49 54 45 ..CSeq: 1 INVITE 0250 0d 0a 46 72 6f 6d 3a 20 22 75 65 41 22 20 3c 73 ..From: "ueA" <s 0260 69 70 3a 75 65 41 40 64 6f 6d 61 69 6e 41 2e 61 ip:ueA@domainA.a 0270 6d 6d 69 2e 64 69 74 2e 75 70 6d 2e 65 73 3e 3b mmi.dit.upm.es>; 0280 74 61 67 3d 31 32 31 32 32 33 34 37 0d 0a 54 6f tag=12122347..To 0290 3a 20 3c 73 69 70 3a 75 65 43 40 64 6f 6d 61 69 : <sip:ueC@domai 02a0 6e 43 2e 61 6d 6d 69 2e 64 69 74 2e 75 70 6d 2e nC.ammi.dit.upm. 02b0 65 73 3e 0d 0a 56 69 61 3a 20 53 49 50 2f 32 2e es>..Via: SIP/2. 02c0 30 2f 55 44 50 20 5b 32 30 30 31 3a 37 32 30 3a 0/UDP [2001:720: 02d0 31 35 30 30 3a 32 34 3a 46 43 46 44 3a 46 46 3a 1500:24:FCFD:FF: 02e0 46 45 30 44 3a 31 30 31 5d 3b 62 72 61 6e 63 68 FE0D:101];branch 02f0 3d 7a 39 68 47 34 62 4b 31 32 30 35 2e 64 64 62 =z9hG4bK1205.ddb 0300 66 38 37 65 34 2e 30 0d 0a 56 69 61 3a 20 53 49 f87e4.0..Via: SI 0310 50 2f 32 2e 30 2f 55 44 50 20 5b 32 30 30 31 3a P/2.0/UDP [2001: 0320 37 32 30 3a 31 35 30 30 3a 31 42 3a 46 43 46 44 720:1500:1B:FCFD 0330 3a 46 46 3a 46 45 30 43 3a 31 30 33 5d 3b 62 72 :FF:FE0C:103];br 0340 61 6e 63 68 3d 7a 39 68 47 34 62 4b 31 32 30 35 anch=z9hG4bK1205 0350 2e 63 38 61 36 36 36 32 36 2e 30 0d 0a 56 69 61 .c8a66626.0..Via 0360 3a 20 53 49 50 2f 32 2e 30 2f 55 44 50 20 5b 32 : SIP/2.0/UDP [2 0370 30 30 31 3a 37 32 30 3a 31 35 30 30 3a 31 42 3a 001:720:1500:1B: 0380 46 43 46 44 3a 46 46 3a 46 45 30 43 3a 31 30 32 FCFD:FF:FE0C:102 0390 5d 3b 62 72 61 6e 63 68 3d 7a 39 68 47 34 62 4b ];branch=z9hG4bK 03a0 31 32 30 35 2e 31 63 62 33 61 30 62 34 2e 30 0d 1205.1cb3a0b4.0. 03b0 0a 56 69 61 3a 20 53 49 50 2f 32 2e 30 2f 55 44 .Via: SIP/2.0/UD 03c0 50 20 5b 32 30 30 31 3a 37 32 30 3a 31 35 30 30 P [2001:720:1500 03d0 3a 31 41 3a 46 43 46 44 3a 46 46 3a 46 45 30 41 :1A:FCFD:FF:FE0A 03e0 3a 31 30 33 5d 3b 62 72 61 6e 63 68 3d 7a 39 68 :103];branch=z9h 03f0 47 34 62 4b 31 32 30 35 2e 34 34 33 34 34 66 30 G4bK1205.44344f0 0400 34 2e 30 0d 0a 56 69 61 3a 20 53 49 50 2f 32 2e 4.0..Via: SIP/2. 0410 30 2f 55 44 50 20 5b 32 30 30 31 3a 37 32 30 3a 0/UDP [2001:720: 0420 31 35 30 30 3a 31 39 3a 46 43 46 44 3a 46 46 3a 1500:19:FCFD:FF: 0430 46 45 30 45 3a 31 30 31 5d 3b 62 72 61 6e 63 68 FE0E:101];branch 0440 3d 7a 39 68 47 34 62 4b 31 32 30 35 2e 61 34 34 =z9hG4bK1205.a44 0450 65 33 62 33 34 2e 30 0d 0a 56 69 61 3a 20 53 49 e3b34.0..Via: SI 0460 50 2f 32 2e 30 2f 55 44 50 20 5b 32 30 30 31 3a P/2.0/UDP [2001: 0470 37 32 30 3a 31 35 30 30 3a 31 61 3a 32 33 30 3a 720:1500:1a:230: 0480 31 62 66 66 3a 66 65 31 30 3a 35 66 62 33 5d 3a 1bff:fe10:5fb3]: 0490 36 31 30 35 32 3b 62 72 61 6e 63 68 3d 7a 39 68 61052;branch=z9h 04a0 47 34 62 4b 34 30 65 31 38 34 34 35 31 32 61 32 G4bK40e1844512a2 04b0 33 34 65 33 32 36 65 66 38 32 32 32 66 38 62 32 34e326ef8222f8b2 04c0 30 34 66 62 0d 0a 4d 61 78 2d 46 6f 72 77 61 72 04fb..Max-Forwar 04d0 64 73 3a 20 31 32 0d 0a 43 6f 6e 74 61 63 74 3a ds: 12..Contact: 04e0 20 22 75 65 41 22 20 3c 73 69 70 3a 5b 32 30 30 "ueA" <sip:[200 04f0 31 3a 37 32 30 3a 31 35 30 30 3a 31 61 3a 32 33 1:720:1500:1a:23 0500 30 3a 31 62 66 66 3a 66 65 31 30 3a 35 66 62 33 0:1bff:fe10:5fb3 0510 5d 3a 36 31 30 35 32 3b 74 72 61 6e 73 70 6f 72 ]:61052;transpor 0520 74 3d 75 64 70 3e 0d 0a 50 2d 50 72 65 66 65 72 t=udp>..P-Prefer 0530 72 65 64 2d 49 64 65 6e 74 69 74 79 3a 20 22 75 red-Identity: "u 0540 65 41 22 20 3c 73 69 70 3a 75 65 41 40 64 6f 6d eA" <sip:ueA@dom 0550 61 69 6e 41 2e 61 6d 6d 69 2e 64 69 74 2e 75 70 ainA.ammi.dit.up 0560 6d 2e 65 73 3e 3b 74 61 67 3d 31 32 31 32 32 33 m.es>;tag=121223 0570 34 37 0d 0a 50 72 69 76 61 63 79 3a 20 6e 6f 6e 47..Privacy: non 0580 65 0d 0a 50 2d 41 63 63 65 73 73 2d 4e 65 74 77 e..P-Access-Netw 0590 6f 72 6b 2d 49 6e 66 6f 3a 20 33 47 50 50 2d 55 ork-Info: 3GPP-U 05a0 54 52 41 4e 2d 54 44 44 3b 75 74 72 61 6e 2d 63 TRAN-TDD;utran-c 05b0 65 6c 6c 2d 69 64 2d 33 67 70 70 3d 31 38 31 30 ell-id-3gpp=1810 05c0 38 36 0d 0a 41 6c 6c 6f 77 3a 20 49 4e 56 49 54 86..Allow: INVIT 05d0 45 2c 41 43 4b 2c 43 41 4e 43 45 4c 2c 42 59 45 E,ACK,CANCEL,BYE 05e0 2c 50 52 41 43 4b 2c 55 ,PRACK,U *********************************************************************** If you want the .cap file for open it with ethereal, ask it for me. Thanks for your help. ______________________________________________ LLama Gratis a cualquier PC del Mundo. Llamadas a fijos y móviles desde 1 céntimo por minuto. http://es.voice.yahoo.com

1 0

Re: [Serusers] Problem with MTU size in SER with MIPv6.
by Jose Angel Calvo 21 Dec '06

21 Dec '06

I'm sorry for the delay, but we had to do a deliverable in this date, and it was imposible to answer before. >Could you try again after runing: >ulimit -c unlimited # you need bash for this >ser -f ser.cfg -w /tmp/ The parameter "unlimited" already was in this node, so I think this is not the problem. The scenario used in our test is the following: - There are two user terminals that are communicated by SIP Communicator program. The INVITE message (problematic paquet) travels through the nodes until it arrive at the Proxy node. In this node is the problem with the core shown in my previous message. We have MIPv6 in our scenario. In our case, we use mipv6-2.0.2 from http://www.mobile-ipv6.org/ . We have ubuntu with kernel 2.6.16.24 Next, i'll show you the configuration file in the Proxy node: ****************************************************************** # # P-CSCF # # ----------- global configuration parameters ------------------------ check_via=no # (cmd. line: -v) dns=yes # (cmd. line: -r) rev_dns=yes # (cmd. line: -R) fifo="/tmp/ser_fifo" listen="pcscf" # ------------------ module loading ---------------------------------- loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/xlog.so" # ----------------- setting module-specific parameters --------------- # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; if (method=="REGISTER") { # Append the Path header # FIXME: "Require" and "Supported" issues append_hf("Path: <sip:term@pcscf.domainA.ammi.dit.upm.es;lr>\r\n"); append_hf("Require: path\r\n"); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (method != "REGISTER") { record_route(); }; if (loose_route()) { route(1); break; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (!t_relay()) { sl_reply_error(); }; } **************************************************************** In the next message i'll show the packet INVITE that produces the error becuase. Otherwise, the message would be very big (>40 KB) and it would not be shown. Thanks. ______________________________________________ LLama Gratis a cualquier PC del Mundo. Llamadas a fijos y móviles desde 1 céntimo por minuto. http://es.voice.yahoo.com

1 0

[Serusers] editing the location table.
by Natambu Obleton 21 Dec '06

21 Dec '06

I want to edit the mysql location table myself, but it seems that SER doesn't read the table everytime a call is received. How can I make it refresh this table? Right now if I stop and restart SER then it will load my new entries. Natambu Obleton Network Engineer FastTrack Communications nobleton(a)fasttrackcomm.net (970) 247-3366 office (970) 247-2426 fax

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[Serusers] Maximum Time of the call
by Digital Tuffy 21 Dec '06

21 Dec '06

How can we restrict a user for the maximum time allowed for a call. _________________________________________________________________ Get up-to-date with movies, music and TV. Its happening on MSN Entertainment http://content.msn.co.in/Entertainment/Default

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RE: [Serusers] editing the location table.
by Digital Tuffy 21 Dec '06

21 Dec '06

use serctl ul command only which will update location table as well as DATA in Memory which is in the form of linked lists etc. >From: "Natambu Obleton" <nobleton(a)fasttrackcomm.net> >To: <serusers(a)iptel.org> >Subject: [Serusers] editing the location table. >Date: Wed, 20 Dec 2006 11:13:01 -0700 > >I want to edit the mysql location table myself, but it seems that SER >doesn't read the table everytime a call is received. How can I make it >refresh this table? Right now if I stop and restart SER then it will load >my >new entries. > > > >Natambu Obleton > >Network Engineer > >FastTrack Communications > >nobleton(a)fasttrackcomm.net > >(970) 247-3366 office > >(970) 247-2426 fax > > > > > >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers _________________________________________________________________ Want to know what your favourite stars are up to? Catch the latest gossip in Bollywood on MSN Entertainment! http://content.msn.co.in/Entertainment/Default

1 0

Re: [Users] Delayed signalling?
by Jiri Kuthan 20 Dec '06

20 Dec '06

At 16:03 14/12/2006, samuel wrote: >It might be due to a DNS query....whenver a request has to be >forwarded to a domain, openSER makes a DNS query to resolv the IP. >During this operation, the child processing the request will not >answer to further incoming messages. Hard to say without more input what the cause may be. Indeed common suspects are DNS (or any other blocking operation, such as dataabse) or timers, but again it is hard to say without more input. If it is DNS or TCP-based blocking, it may be worthwhile trying genuine SER/ottendorf which is having IP blacklisting and DNS cachine in there -- this is indeed something which has been pressing us for a while. If it is timer system, then SER should address it it (it must be the latest ottendorf version though). If it is database that blocks itself and SER consequently too, you have to look at the database. (Even if SER would do something about it, it could do hardly better then sending a negative reply "DB unavailable"). -jiri >it also can be happening due to a spiral loop that stays on the server. > >Without further information (confg,logs) it's hard to tell which is >the reason... > >hope it helps, >Samuel. > >without more information > >2006/12/14, Max Gregorian <gregorian442(a)googlemail.com>: >>Hi all, >> >>Just wondering if anyone else has had this problem. I have noticed while >>tracing on my OpenSER server, that every now and then the server receives a >>packet which it does to respond to immediately, resulting in a string of >>packets being sent to the server and then the server responding a few >>seconds later. This does not happen all the time, just say maybe once or >>twice every hour. The rest of the time the signaling is correct and >>responses follow request packets in the correct order. >> >>What I am trying to figure out is whether this is a load traffic issue (i.e. >>can the server not handle too much load), and if so is it OpenSER or the >>network or the server in general? I have run diagnostics on the servers and >>there is nothing wrong with the hardware. >> >>On the other hand Could this be related to any timer issues? I remember >>there was mention of timers in SER but are there any default timer settings >>that can be tweaked? >> >>Thanks in advance for any response. >> >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users -- Jiri Kuthan http://iptel.org/~jiri/

2 2

[Users] OT: network debugging
by Klaus Darilion 20 Dec '06

20 Dec '06

Hi! As SIP debugging often includes low-level network debugging I think this tutorial can give some valuable inputs how to debug network problems. http://penguin-it.co.il/presentations/BrainShare2006/Troubleshooting%20LAN%… regards klaus -- Klaus Darilion nic.at

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[Serusers] Recommend ATA adaptor ?
by johnk＠koreanet.us 20 Dec '06

20 Dec '06

Hi, Can someone recommend ATA models that work easy with Openser, based on experience? I am currently using Sipura SPA3000 and PAP2 which are highly configurable but at the same time, need much works with trial and error fashion to get them work with Openser, Freeradius, Mediaproxy and CDRTool combination. Especially, I am not quite sure if these are handling "VIA" header correctly. Because setting up Outbound proxy was the only solution I found to get these units send "BYE" to the server, otherwise it will send "BYE" directly to PSTN gateway, resulting forward_reply error, killing a child process and mediaproxy never had a chance to mark "stop time" for acc. Can anyone suggest ATAs that imposes least burden of configuration to work with the setting above? Thanks mush in advance. John K.

3 2

[Users] New OpenSER Module: The SNMPStats Module
by jmagder 20 Dec '06

20 Dec '06

Hello everybody, I recently submitted a new module known as the "SNMPStats module" to the OpenSER CVS head. The SNMPStats module provides an SNMP management interface to OpenSER, and is intended for those who want better manageability of their systems. The module provides general SNMP queryable scalar statistics (useful for discovering system trends), table representations of more complicated data such as user and contact information, and alarm monitoring capabilities. The scalar statistics are very useful for discovering trends in the ways your systems are used. For example, using a tool such as mrtg or Cacti, you can collect SNMP based information over long periods of time, and automatically graph statistics. Some examples of good graphs would be: - The number of messages received each hour of the day. (Using the openserSIPSummaryInRequests scalar) - The number of active dialogs at any time of the day. (Using the openserCurNumDialogs scalar) These graphs may reveal that most SIP traffic happens between say, 14:00-16:00, and that perhaps it would be beneficial to change rate plans based on this data. SNMPStats alarm monitoring provides for a useful tool in maintaining system stability. Currently there are two alarms implemented: - Dialog Alarms: If the number of active dialogs passes a certain threshold, then a message can be sent to a network operation center. The idea is that an administrator should me made aware a system is becoming too heavily loaded before a failure occurs. - Message Queue Alarms: If the number of bytes waiting to be consumed by an OpenSER instance exceeds a threshold, then a message can be sent to a network operation center. The idea is to find out why the system is not able to handle the load, and to take corrective actions before a failure occurs. In order to get the SNMPStats module up and running, you'll be required to do some extra configuration steps that at this time are not automated. The full installation procedure is documented on the OpenSER web site at: http://www.openser.org/docs/modules/1.2.x/SNMPStats.html Since this is the first time the SNMPStats module will be tried outside of my work at SOMA Networks, there may be a few rough edges I've overlooked. I welcome all to take out their sand paper help me polish the module. All suggestions and criticisms are always welcome. Good luck all, and cheers! - Jeffrey Magder - SOMA Networks Inc.

3 3

[Users] Trying to understand difference between forward() and t_relay()
by Juan Carlos Castro y Castro 20 Dec '06

20 Dec '06

I was googling for that and found a reply in the SER list that links to a page that doesn't exist anymore: http://lists.iptel.org/pipermail/serusers/2004-October/011924.html My searchs have been unsuccesful so far. Can anyone give me a pointer? Thanks, Juan

2 2

[Users] postgres
by Mark Price 20 Dec '06

20 Dec '06

Hi, What is the best way to use postgres with openser? Is postgres a viable option? I ask, because the install script appears to be broken. If I am able to fix it well enough create the necessary tables, will it be fit for use? As an example, there are several errors in the openser_postgres.sh that comes with openser 1.1 In particular, there are instances where the following is done: create table a ( field type, index a_idx(field) ); However, postgres requires syntax like this: create table a ( field type ); create index a_idx on a (field); Thanks.

2 1

[Serusers] How to change charset in SER for IM
by Killer 20 Dec '06

20 Dec '06

Hi ! Can someone tell how to change the charset in SER ? I want to send IMs which also shows german letters like "ß" ... I also get these Errors and even google couldnt help me :-( , how to erase these erros WARNING: run_actions: null action list (rec_level=1) 2(4154) ERROR: fifo_server: command must have at least 3 chars 2(4154) ERROR: fifo_server: command must have at least 3 chars Thanks a lot for your answers Ahsen ___________________________________________________________ Der frühe Vogel fängt den Wurm. Hier gelangen Sie zum neuen Yahoo! Mail: http://mail.yahoo.de

2 1

[Users] Mysql reconnect problem
by Dimo 20 Dec '06

20 Dec '06

Hi All, I have a problem with openser reconnecting to my redundant mysql servers which hold the openser database. The database servers are in a cluster and if one fails the other takes over its IP address. In openser however the connection to the database is established only (i think) at the beginning. So when openser is running and one of my sql servers drops, and the other takes over the IP, i get an error in openser saying that the mysql connection does not exist. Is there a way to make openser reconnect to mysql automatically without restarting it or some other way I can make this work? Best, Dimo

4 6

[Serusers] cannot find directory /src/sip_router/tools/serct
by Franz-Ferdinand Lehnert 20 Dec '06

20 Dec '06

Hello fellows I was looking for the 'magic' scripts (http://ftp.iptel.org/pub/ser/presence/presence-handbook/ar01s07.html) to initialize the postgres data base for ser 0.10xxx (latest snapshot from today) quite some time, but could not find the folder 'src/sip_router/tools' to use the new serctl. Even my search with google gave no result. Can someone help me please? Kind regards leffe

3 3

Re: [Serusers] Help! Run SER with MySQL to do authentication
by Greger V. Teigre 20 Dec '06

20 Dec '06

Copy the list. Well, "libmysqlclient.so.14: cannot open shared object file: No such file or directory" pretty much says it all, right? Your LD_LIBRARY_PATH environment variable must include your library dirs before starting ser (cumbersome) or you need to add /usr/lib/mysql to your /etc/ld.so.conf file (which btw should have been updated when installing mysql, I don't know why it didn't happen for you). g-) Tako Chang wrote: > Thanks!! > I ran make_and_install script > But i still can't run 'auth-mysql.cfg' > Error message : > load_module: could not open module > </usr/local/lib/ser/modules/mysql.so>: libmysqlclient.so.14: cannot > open shared object file: No such file or directory > > but I have libmysqlclient.so.14 at "/usr/local/lib/mysql" > What should I do?? > > Thanks for your help <(_ _)> > > I mentioned that someone talked about the version of mysql > So, i download the older version 4.1.22 > 2006/12/6, Greger V. Teigre < greger(a)teigre.com > <mailto:greger@teigre.com>>: > > Look at the make_and_install script in the scripts package (same > place). It does this for you. mysql-devel must be installed. > g-) > > Tako Chang wrote: >> Hi, all >> >> First of all, I read [SER - Getting Started >> <http://siprouter.onsip.org/doc/gettingstarted/>] and downloaded >> [ SER Source package >> <ftp://siprouter.onsip.org/pub/gettingstarted/packages/ser-0.9.7-pre1.Gettin…>] >> >> after I make the package under debug mode, it runs well. >> Now, I can receive REGISTER requests and send 200 OK Responses >> and play a SIP proxy server >> >> I want to test "auth_mysql.cfg" in Chapter 7.... so, >> i remove "exclude_modules?= jabber mysql\" in Makefile >> but i still can't compile the mysql.so >> >> Is there some more detail documents or suggestions for me? >> >> Thank! >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org <mailto:Serusers@lists.iptel.org> >> http://lists.iptel.org/mailman/listinfo/serusers >> > >

2 2

[Users] (PR)ACK problem with uac_replace_from
by Jeff Williams 20 Dec '06

20 Dec '06

I seem to have an issue with uac_replace_from. I am using openser as a media proxy. For invites I replace the numbers with leading 0's with 61 for Australia and forward the call to our LCR box which then forwards the call to the correct voice to pstn gateway. I am trying to use uac_replace_from to set the from address so the callerid appears correct on the outgoing calls. This seems to work fine for the INVITE request, but on subsequent PRACKs and ACKs (which just get loose routed), something strange happens to the from address. On invite, the From get re-written: From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1790577941 to From: Jeff <sip:0892209080@sipone.com>;tag=1790577941 For the PRACKs and ACKs, this happens: From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1790577941 to From: Jeff <\032\317\352S\232\222B\230\234K\231*BV\300\030\303\332\ah\303\306i\365\005\016\200\250\000\000\230\016\200\252j\200\002`\000\000\016>;tag=1790577 941 There shouldn't be anything happening to the From address on the (PR)ACKs at all. Anyone have any ideas? I have attached a ethereal dump of the call setup. Jeff

2 2

[Users] RE: [Serusers] Recommend ATA adaptor ?
by johnk＠koreanet.us 19 Dec '06

19 Dec '06

Thanks for quick reponse. Actually I agree with you fully but the only reason for looking at other choice is that I ran out of ideas to resolve this Sipura "hang up" situations where OpenSER logs "forward_reply" error and Mediaproxy's missing stop-time for the call. The only way of getting around was to setup Sipura to use Outbound proxy of same server, which I dearly want to avoid in production. Any idea how to resolve the issue? > You won't find anything better than Sipura/ Linksys. I tried a lot and it > is > worth it to make a script to configure instead of looking for others. > There > is a lot of trash around also. > > Regards CFM > > -----Original Message----- > From: serusers-bounces(a)lists.iptel.org > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of johnk(a)koreanet.us > Sent: Tuesday, December 19, 2006 3:04 PM > To: serusers(a)iptel.org > Subject: [Serusers] Recommend ATA adaptor ? > > Hi, > > Can someone recommend ATA models that work easy with Openser, based on > experience? > > I am currently using Sipura SPA3000 and PAP2 which are highly configurable > but at the same time, need much works with trial and error fashion to get > them work with Openser, Freeradius, Mediaproxy and CDRTool combination. > > Especially, I am not quite sure if these are handling "VIA" header > correctly. Because setting up Outbound proxy was the only solution I > found > to get these units send "BYE" to the server, otherwise it will send "BYE" > directly to PSTN gateway, resulting forward_reply error, killing a child > process and mediaproxy never had a chance to mark "stop time" for acc. > > Can anyone suggest ATAs that imposes least burden of configuration to work > with the setting above? > > Thanks mush in advance. > > John K. > > > > > > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > -- > No virus found in this incoming message. > Checked by AVG Free Edition. > Version: 7.5.432 / Virus Database: 268.15.24/592 - Release Date: > 12/18/2006 > 1:45 PM > > > -- > No virus found in this outgoing message. > Checked by AVG Free Edition. > Version: 7.5.432 / Virus Database: 268.15.24/592 - Release Date: > 12/18/2006 > 1:45 PM > > >

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[Users] Recommend ATA adaptor ?
by johnk＠koreanet.us 19 Dec '06

19 Dec '06

Hi, Can someone recommend ATA model that works easy with Openser, based on experience? I am currently using Sipura SPA3000 and PAP2 which are highly configurable but at the same time, need much works with trial and error fashion to get them work with Openser, Freeradius, Mediaproxy and CDRTool combination. Especially, I am not quite sure if these are handling "VIA" header correctly. Because setting up Outbound proxy was the only solution I found to get these units send "BYE" to the server, otherwise it will send "BYE" directly to PSTN gateway, resulting forward_reply error, killing a child process and mediaproxy never had a chance to mark "stop time" for acc. Can anyone suggest ATAs that imposes least burden of configuration to work with the setting above? Thanks mush in advance. John K.

1 0

[Users] re: how to pass the Caller ID into uac_replace_from function ?
by don lin 19 Dec '06

19 Dec '06

Hi there, We have a Postgres Database to store the users' informations before we use Openser. To integrate between Postgres and Openser, we write external C programs and call them from Openser with exec_dset function. The issue we have now is with the Caller ID. When the user makes a call to PSTN number from SIP devices (soft phone or hard phone), the telco carrier requests us to send a numerical Caller ID, not the SIP format (such as john.dow(a)abc.com). What I am doing is to add the Caller ID field into the Postgres for each user and write a C program to query the Caller ID field, then Openser calls it with exec_dset functioin. However, I did not find a way to define a variable in Openser to store this Caller ID and pass it to uac_replace_from function. Appreciate for any suggestions. Jeff, You mentioned that you replaced the From Header, may I ask how you do it ? Thanks and Regards, Don ------------------------------ <Message: 4 <Date: Tue, 19 Dec 2006 11:11:09 +0900 <From: Jeff Williams <jeffw(a)globaldial.com> <Subject: [Users] (PR)ACK problem with uac_replace_from <To: users(a)openser.org <Message-ID: <45874A3D.5070206(a)globaldial.com> <Content-Type: text/plain; charset="iso-8859-1" <I seem to have an issue with uac_replace_from. <I am using openser as a media proxy. For invites I replace the numbers <with leading 0's with 61 for Australia and forward the call to our LCR <box which then forwards the call to the correct voice to pstn gateway. <I am trying to use uac_replace_from to set the from address so the <callerid appears correct on the outgoing calls. <This seems to work fine for the INVITE request, but on subsequent PRACKs <and ACKs (which just get loose routed), something strange happens to <the from address. <On invite, the From get re-written: <From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1790577941 <to <From: Jeff <sip:0892209080@sipone.com>;tag=1790577941 <For the PRACKs and ACKs, this happens: <From: Jeff <sip:610089001@proxy.sipone.com;user=phone>;tag=1790577941 <to <From: Jeff <<\032\317\352S\232\222B\230\234K\231*BV\300\030\303\332\ah\303\306i\365\005\016\200\250\000\000\230\016\200\252j\200\002`\000\000\016>;tag=1790577 <941 <There shouldn't be anything happening to the From address on the <(PR)ACKs at all. <Anyone have any ideas? <I have attached a ethereal dump of the call setup. <Jeff __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Users] How can we do Calling Card System with OpenSER
by raviprakash sunkara 19 Dec '06

19 Dec '06

Hello Users First of All Advance "Marry Chrimas " Is How can we do the Calling Card System with OpenSER, I don't want the route every call to Asterisk ( PBX feature ) and other How can we perform the Billing System in OpenSER for Post Paid anf PrePaid Billing Systeem In Asterisk it can do it... by thrid party open Source .... My main intension is with OpenSER with Calling Card System -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 ravi.sunkara(a)hyperion-tech.com www.hyperion-tech.com

2 1

[Users] ACK after INVITE not reach callee
by somsak vattanavakin 19 Dec '06

19 Dec '06

Dear List if ACK message after INVITE message (UAC caller ) can not send to UAC callee. can i do something on openser to complete this 3 way handshaking. openser server bedind NAT best Somsak Vattanavakin 2753-7 Ladprao 99-101 Bangkok.10310 Thailand Tel. (66)0-29350016 , 0-25386251 Fax. (66)0-25140253 www.infowavenet.com

2 1

[Users] client_nat_test
by Andreas Granig 19 Dec '06

19 Dec '06

Hi, Provided that my clients are advised to not use STUN or any other NAT traversal mechanism, is it enough to just use client_nat_test("2"), which checks if source-ip is different from the one in the Via-Header? Should be sufficient for detecting NAT, right? Andy

6 16

[Users] Install SER
by Danilo Vilela Dantas 19 Dec '06

19 Dec '06

Hi How does it install SER 0.9.6 in the Debian 3.1? And your modules? Througth apt-get tool? Thanks! Danilo

3 2

[Users] openser-1.1.0-notls not compile in NetBSD
by Roberto Pereyra 19 Dec '06

19 Dec '06

Hi all ! I trying to complie openser-1.1.0-notls in my netbsd host and not compile. My system is: NetBSD 3.1_RC4 Thanks for any help !! roberto My output is: smtp2: {6} gmake mode=debug all Compiling cfg.tab.c gcc -DNAME='"openser"' -DVERSION='"1.1.0-notls"' -DARCH='"i386"' -DOS='"netbsd"' -DCOMPILER='"gcc 3.3.3"' -D__CPU_i386 -D__OS_netbsd -D__SMP_no -DCFG_DIR='"/usr/pkg/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DF_MALLOC -DSTATISTICS -DEXTRA_DEBUG -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_SOCKADDR_SA_LEN -DHAVE_GETHOSTBYNAME2 -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_CONNECT_ECONNRESET_BUG -DHAVE_TIMEGM -DHAVE_SELECT -c cfg.tab.c -o cfg.tab.o cfg.y: In function `yyparse': cfg.y:859: error: `IPTOS_MINCOST' undeclared (first use in this function) cfg.y:859: error: (Each undeclared identifier is reported only once cfg.y:859: error: for each function it appears in.) gmake: *** [cfg.tab.o] Error 1 smtp2: {7} -- Ing. Roberto Pereyra ContenidosOnline Looking for Linux Virtual Private Servers ? Click here: http://www.spry.com/hosting-affiliate/scripts/t.php?a_aid=426&a_bid=56

1 0

[Users] How to get CDR in Openser
by Howard Tang 19 Dec '06

19 Dec '06

Hi, Can someone give me some idea of how i can obtain CDR from Openser please? I saw a open source software called cdrtools, however, i don't want to install apache on my machine. Any other CDR software will work with openser? Regards, Howard

2 1

[Serusers] Missed calls in serweb
by hiSIPatH 19 Dec '06

19 Dec '06

Hi all, i have ser-0.9.6 and serweb (0.9.6), with mysql support and SQL accounting enabled (DEFS+=-DSQL_ACC). Unfortunately i am not able to configure missed calls correctely (not being displayed in serweb and no entry in mysql). mysql> select * from missed_calls; Empty set (0.01 sec) what's wrong with my config? thx a lot... my config looks like: [...] loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/acc.so" modparam("auth_db|acc|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "log_missed_flag", 3 ) modparam("acc", "db_missed_flag", 3) modparam("acc", "db_flag", 1) modparam("acc", "log_level", 1) route{ [....] if (!lookup("location")) { if ((method=="INVITE" || method=="ACK") && t_newtran() ) { t_reply("404", "Not Found"); acc_log_request("404 Not Found"); acc_db_request("404 missed call", "missed_calls"); break; }; sl_send_reply("404", "User Not Found"); break; }else { setflag(3); t_relay(); break; };

1 0

[Users] avp concat
by vallimamod abdullah 19 Dec '06

19 Dec '06

Hello list, I have 2 avps containing a username ($avp(s:usr))and a domain ($avp(s:domain)) retrieved from db and I would like to build an uri by joining this 2 avps with an '@' in the middle. I have checked avp_subst and avp_printf but it doesn't work as, for what I have understood, they only accept pseudo variables (and not other avps) for the replacement. For example, avp_printf("$avp(s:uri)", "$avp(s:usr)@$avp(s:domain)") is not valid. Is there a (simple) way to achieve what I want (I am using version 1.1.0-notls) ? Thank you, -vma .

2 3

[Users] How to get statistics from the dialog module.
by Leonid Fainshtein 19 Dec '06

19 Dec '06

Hi, I can't get dialog:active_dialogs statistic in my custom module. There is a fragment of my code: stat_var *psvActive_dlg_stat; str sActive_dlg; long lCurrentNumActiveDlg =0; char szStatisticName[ 25]; strcpy( szStatisticName, "dialog:active_dialogs"); sActive_dlg.s = szStatisticName; sActive_dlg.len = strlen( szStatisticName); psvActive_dlg_stat = get_stat( &sActive_dlg ); if (psvActive_dlg_stat) lCurrentNumActiveDlg = get_stat_val( psvActive_dlg_stat ); .... The problem is that get_stat() always returns 0. Also I get error when I try to get the statistic by using openserctl: # openserctl fifo get_statistics dialog:active_dialogs 404 Statistic not found But the following works ok: ClaroSIS22:/usr/local/sbin# openserctl fifo get_statistics dialog: 200 OK Module name = dialog; statistics=3 dialog:active_dialogs = 0 dialog:processed_dialogs = 0 dialog:expired_dialogs = 0 What could be the problem? Thank you in advance, Leonid Fainshtein ******************************************** This outgoing mail message was scanned by McAfee GroupShield Engine %tn%

2 1

[Serusers] append_rpid_hf() replacement in CVS-HEAD
by Hendrik Scholz 18 Dec '06

18 Dec '06

Hi! append_rpid_hf() was removed from CVS in 2005 with the following note: "Rpid support dropped, it will be implemented using AVPs." Is there a standard way of implementing this? OpenSER utilizes avp_printf() for this which isn't part of SER: http://www.mail-archive.com/devel@openser.org/msg02248.html What is the Best Current Practice to add the Rpid header? If there is none I'd help implementing something new since I'm already tackling some Rpid to P-Asserted-Identity transformation things. It would also be handy to add the privacy tag according to another AVP (although I'm not using it). Cheers, Hendrik -- freenet Cityline GmbH, Hamburger Chaussee 2-4, 24114 Kiel, Germany Phone: +49 (0)431 9020552, Fax: +49 (0)431 9020559 Internet: http://www.freenet.de, eMail: hendrik.scholz(a)freenet-ag.de

2 2

[Users] Delayed signalling?
by Max Gregorian 18 Dec '06

18 Dec '06

Hi all, Just wondering if anyone else has had this problem. I have noticed while tracing on my OpenSER server, that every now and then the server receives a packet which it does to respond to immediately, resulting in a string of packets being sent to the server and then the server responding a few seconds later. This does not happen all the time, just say maybe once or twice every hour. The rest of the time the signaling is correct and responses follow request packets in the correct order. What I am trying to figure out is whether this is a load traffic issue (i.e. can the server not handle too much load), and if so is it OpenSER or the network or the server in general? I have run diagnostics on the servers and there is nothing wrong with the hardware. On the other hand Could this be related to any timer issues? I remember there was mention of timers in SER but are there any default timer settings that can be tweaked? Thanks in advance for any response.

7 12

[Serusers] SIPSAK error in username
by "tzieleniewski" 18 Dec '06

18 Dec '06

Hi! I was testing my ser proxy with sipsak and i found one error. invoking sipsak with the following command: # sipsak -IU -vvv -s sip:sipsak@scscf.ims.touk.pl -l 5061 -H 192.168.0.138 -e 1 -S -a sipsak sipsak inserts an extratrailing zero in the username this is both in the to and from headers as well in the authentication header. U 2006/12/18 12:52:48.594546 192.168.0.138:5061 -> 192.168.0.74:5060 REGISTER sip:scscf.ims.touk.pl SIP/2.0. Authorization: Digest username="sipsak0", uri="sip:scscf.ims.touk.pl", algorithm=MD5, realm="scscf.ims.touk.pl", nonce="4586820caf30028588ff88825a90f86abbdb598d", response="66968ba65014ad848a3fbe0cc9331daf". Via: SIP/2.0/UDP 192.168.0.138:5061;branch=z9hG4bK.5fcbc0b0;rport;alias. From: sip:sipsak0@scscf.ims.touk.pl;tag=391d3b1a. To: sip:sipsak0@scscf.ims.touk.pl. Call-ID: 958216986(a)192.168.0.138. CSeq: 2 REGISTER. Content-Length: 0. Max-Forwards: 70. User-Agent: sipsak 0.9.6. Expires: 15. Contact: sip:sipsak0@192.168.0.138:5061. Ive put the whole log in the attached file. Bests Tomasz

2 1

[Users] The usrloc table, Oracle, and Asterisk
by Juan Carlos Castro y Castro 18 Dec '06

18 Dec '06

I'm working on an Asterisk+OpenSER setup and I am in the process of implementing the following: registration has to be handled by OpenSER, but it has to be recognized by Asterisk. For that purpose, OpenSER is authenticating against Asterisk's "sipfriends" realtime table. That works OK. But OpenSER should also update fields "ipaddr", "port", and "regseconds" in the same table, or else Asterisk won't be able to direct calls to the logged SIP phones. I'm thinking of implementing an Oracle trigger (or rather asking the Oracle DBA to do that) so all updates/inserts into the location table will cause updates in sipfriends. As far as I know, I need to take the IP and port from the "contact" field of the location table. Or should I use the "received" field? In my tests, the "received" field gets set to NULL, although "contact" has the info I need. Can I use the data from "contact" always? For the regseconds field, I'll just have to make a conversion from the local time format (used by the "expires" field of location) to the unix timestamp format (used by "regseconds" in Asterisk). Oh, by the way, this is the location table definition that worked OK with Oracle. I had to ask for a login trigger so the date fields would accept the 'YYYY-MM-DD HH:MM:SS' format. Also, Oracle refuses to accept an empty string as a non-null value. CREATE TABLE location ( username varchar2(64) DEFAULT '', domain varchar2(128) DEFAULT 'vono.net.br', contact varchar2(255) DEFAULT '', received varchar2(255) DEFAULT '', path varchar2(255) DEFAULT '', expires date DEFAULT '2020-05-28 21:32:15' NOT NULL, q decimal(10,2) DEFAULT 1.0 NOT NULL, callid varchar2(255) DEFAULT 'Default-Call-ID' NOT NULL, cseq int DEFAULT 42 NOT NULL, last_modified date DEFAULT '1900-01-01 00:00' NOT NULL, flags int DEFAULT 0 NOT NULL, user_agent varchar2(255) DEFAULT '', socket varchar2(128) DEFAULT '', methods int DEFAULT NULL, PRIMARY KEY(username, domain, contact) );

7 24

[Users] Is there any module for P&IM?
by Danilo Vilela Dantas 18 Dec '06

18 Dec '06

Hi all, Congratulations! I use Openser. I have got a question. Has it Instant Messaging? It works like IM? Is there any module for P&IM? Thanks! Danilo

3 2

[Users] calculating phplib_id
by Bill Neely 18 Dec '06

18 Dec '06

Can anybody explain to me how to calculate phplib_id? I have been aboe to calculate ha1 and ha1b, but don't see any instructions on calculating the phplib_id. --

2 1

[Users] OpenSER
by Danilo Vilela Dantas 18 Dec '06

18 Dec '06

Hi all What is version OpenSER for the module IMC? Can it use in the OpenSER 1.0.1? And when will be ready the version 1.2.0? Delay? When time delay? Danilo

2 1

Re: [Serusers] branch parameter
by Tuffy Dhawan 18 Dec '06

18 Dec '06

www.adoreinfotech.com is offering a Softphone based on Microsoft RTC/1.2 which is not generating branch parameter while sending INVITE but call is still going through. How is branch parameter used as per RFC 3261. >From: "Pat Wang" <wangyu39(a)hotmail.com> >To: "Kapil Dhawan" <digitaltuffy(a)hotmail.com>,<serusers(a)lists.iptel.org> >Subject: Re: [Serusers] branch parameter >Date: Mon, 18 Dec 2006 00:17:26 -0000 > >branch in Via header is a MUST have parameter by RFC 3261. > >----- Original Message ----- From: "Kapil Dhawan" ><digitaltuffy(a)hotmail.com> >To: <serusers(a)lists.iptel.org> >Sent: Sunday, December 17, 2006 4:05 PM >Subject: [Serusers] branch parameter > > >>Whats the importance of branch parameter. I am using RTC/1.2 library and >>its not generating branch parameter. >> >>_________________________________________________________________ >>Tried the new MSN Messenger? Its cool! Download now. >>http://messenger.msn.com/Download/Default.aspx?mkt=en-in >> >>_______________________________________________ >>Serusers mailing list >>Serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers >> > _________________________________________________________________ Stay in tune with the latest styles! Check out the happening fashion trends on 'Wardrobe watch'. Only on MSN Lifestyle http://content.msn.co.in/Lifestyle/Moreonlifestyle/LifeStyle_091006_0551.htm

2 1

[Serusers] Denial-of_service
by Tuffy Dhawan 18 Dec '06

18 Dec '06

How do you prevent Denial-Of-service in case of a stateful Proxy. _________________________________________________________________ Catch all the cricketing action right here. Live score, match reports, photos et al. http://content.msn.co.in/Sports/Cricket/Default.aspx

1 0

[Serusers] Request Response Query
by Kapil Dhawan 17 Dec '06

17 Dec '06

As per RFC 3261, SIP Trapezoid UAC1 - INVITE - UAS1 UAS1 - INVITE - UAS2 UAS1 - 100 Trying - UAC1 But i tried to check logs in SER which shows UAC1 - INVITE - UAS1 UAS1 - 100 Trying - UAC1 UAS1 - INVITE - UAS2 Why is it different. _________________________________________________________________ Mega Fare sale on domestic airlines. Call 1-800-11-8747 now! http://india.makemytrip.com/?Camp=MSN&attrib=Dec/Tag/Mega

1 0

Re: [Users] Duplicate INVITEs - t_lookup_request and t_check_trans
by Mahesh Paolini-Subramanya 16 Dec '06

16 Dec '06

quite quite positive. the reason i am so certain is that there is only *one* place in my entire config file that i actually do a t_relay() (or equivalent). I isolated all the code into route[1], to ease maintainability of my (admittedly complex) config file. In route[1], the relevant section is pretty damn close to if (!t_relay()) { if (!is_method("ACK")) { sl_reply_error(); } } fwiw, its 1.1.0 (latest cvs version thereof) cheers ----- Original Message ----- From: Bogdan-Andrei Iancu < bogdan(a)voice-system.ro > To: mahesh(a)aptela.com Cc: Klaus Darilion < klaus.mailinglists(a)pernau.at >, users < users(a)openser.org > Sent: Thursday, December 14, 2006 12:39:04 PM GMT-0600 Subject: Re: [Users] Duplicate INVITEs - t_lookup_request and t_check_trans Hi Mahesh, I do not think you need t_check_tran() as t_relay() automatically absorbs the retransmissions. are you sure you are not calling t_relay() twice for the same requests? or calling a tm function (t_newtran) before the t_relay ?? regards, bogdan Mahesh Paolini-Subramanya wrote: >Well, i read the documentation again, and found t_lookup_request, and t_check_trans. >Question 1 - t_lookup_request() creates the transaction if it doesn't exist, correct? However, t_check_trans only checks to see if the transaction already exists, correct? > >Question 2 - Assuming that t_check_trans only checks to see if the transaction already exists, then is the following correct? > >Currently I do the following (after stripping out a bunch of stuff) > >if (!t_relay()) { > if (!is_method("ACK")) { > sl_reply_error(); > } >} > >should this, instead, actually look like the following? >if (is_method("INVITE")) { > if (if !t_check_tran()) { > t_relay(); > } >} else if (!is_method("ACK")) { > sl_reply_error(); >} > > > >----- Original Message ----- >From: Klaus Darilion < klaus.mailinglists(a)pernau.at > >To: mahesh(a)aptela.com >Cc: users < users(a)openser.org > >Sent: Thursday, December 14, 2006 2:06:01 AM GMT-0600 >Subject: Re: [Users] Duplicate INVITEs > >Mahesh Paolini-Subramanya wrote: > > >>We're seeing sporadic situations where fones >>a) send an INVITE >>b) ignore the 100-Trying response, and instead >>c) send another INVITE with the same sequence number. >> >>The question is, What should happen here? >> >> > >If the message is completely identical then it is a retransmission. >Retransmissions should be detected by tm module (e.g. t_relay()) and >absorbed, > >regards >klaus > > > > >>Currently, we send back a 500 response (server error). >>Is this correct? (i think not, cause it invariably causes the fone to go fast busy). >>Is this some other response that should occur? >> >>poring over 3261 resulted in a headache, and no additional clarity... >> >> >> >> >> >> >> >>------------------------------------------------------------------------ >> >>_______________________________________________ >>Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > > > -- ******************************************* Mahesh Paolini-Subramanya (703) 386-1500 x9100 CTO mahesh(a)aptela.com Aptela, Inc. http://www.aptela.com "Aptela: How Business Answers The Call" ******************************************* -- ******************************************* Mahesh Paolini-Subramanya (703) 386-1500 x9100 CTO mahesh(a)aptela.com Aptela, Inc. http://www.aptela.com "Aptela: How Business Answers The Call" *******************************************

1 0

Re: usrloc loading (was Re: [Serusers] Re: Fw: [Users] TM : retransmission timers
by Jiri Kuthan 16 Dec '06

16 Dec '06

Thanks, how could I have missed that! Is what it does it loads usrloc table piece-wise? And if so, why does it do only for mysql? And also, it is saying that old version would have increased memory consumption with TCP/TLS -- does it mean that it does not increase with the new version? -jiri At 14:32 30/11/2006, Ovidiu Sas wrote: >Maybe you are looking for this one: >http://www.openser.org/index.php?option=com_content&task=view&id=48&Itemid=9 > > >Regards, >Ovidiu Sas > >On 11/30/06, Jiri Kuthan <jiri(a)iptel.org> wrote: >>At 12:53 22/11/2006, Weiter Leiter wrote: >>>I know that OpenSER loads (only?) faster. >> >>Can folks share with me what the fast-usrloc-loading feature is about? >>I was not successful finding it out. >> >>Thanks! >> >>-jiri >> >> >>-- >>Jiri Kuthan http://iptel.org/~jiri/ >> >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users > >-- >Jiri Kuthan http://iptel.org/~jiri/

10 18

[Users] determining if From==To
by Mark Price 15 Dec '06

15 Dec '06

Hi, How would one determine if the From address equals the To address, or even better, if the To address is either the same as or an alias of the From address? Thanks Mark Price

4 5

RES: [Users] OpenSER
by Danilo Vilela Dantas 15 Dec '06

15 Dec '06

List of contacts and instant message. ________________________________ De: Adrian Georgescu [mailto:ag@ag-projects.com] Enviada em: sexta-feira, 15 de dezembro de 2006 17:16 Para: Danilo Vilela Dantas Cc: users(a)openser.org Assunto: Re: [Users] OpenSER Hi Danilo, Could you please tell us what features from 1.2.0 do you need so badly? Adrian On Dec 15, 2006, at 8:05 PM, Danilo Vilela Dantas wrote: Hi all What is version OpenSER for the module IMC? Can it use in the OpenSER 1.0.1? And when will be ready the version 1.2.0? Delay? When time delay? Danilo _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

1 0

[Users] new MediaProxy release 1.8.0
by Adrian Georgescu 15 Dec '06

15 Dec '06

Hello, There is a new MediaProxy release available. To upgrade go to http://mediaproxy.ag-projects.com or download the software directly from: http://mediaproxy.ag-projects.com/mediaproxy-1.8.0.tar.gz Changes from version 1.7.2 to 1.8.0 ----------------------------------- - Added support for radius accounting. The accounting type can be specified in the configuration file. To support radius accounting the pyrad module must be installed (available at http://www.wiggy.net/code/pyrad/). - The old boolead switch dbaccounting in the [Accounting] section of mediaproxy.ini was replaced by accounting which is a string specifying the accounting type to be used: none, radius or database. For those using CDRTool, release 5.0 is available, which works with the new Radius accounting of MediaProxy 1.8.0. To upgrade go to http://cdrtool.ag-projects.com/ or download the software directly from: http://download.dns-hosting.info/CDRTool/cdrtool_5.0.tar.gz Kind regards, Adrian Georgescu

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[Users] NEW FEATURE: new pseudo-variables from dialog module
by Bogdan-Andrei Iancu 15 Dec '06

15 Dec '06

Hi everybody, the dialog module exports 2 new pseudo variables: *DLG_lifetime* - the duration (in seconds) of the dialog. The duration is calculated from the dialog confirmation and the current moment. This PV will be available only for sequential requests, after doing loose_route(). So, it can be used to get timestamps foe sequential requests, or to get the overall dialog duration and to use it for accounting. Closes Feature Request 1611930. https://sourceforge.net/tracker/index.php?func=detail&aid=1611930&group_id=… *DLG_status* - Returns the status of the dialog. This PV will be available only for sequential requests, after doing loose_route(). Value may be: 3 - Confirmed by a final reply but no ACK received yet. 4 - Confirmed by a final reply and ACK received. 5 - Dialog ended. Can be used to delay re-INVITEs until ACK is received and avoid signalling races. Closes Feature Request 1610630. https://sourceforge.net/tracker/index.php?func=detail&aid=1610630&group_id=… regards, bogdan

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[Serusers] Interdomain remote disconnect
by Maciej Żwirski 15 Dec '06

15 Dec '06

Hi, I've a following problem: I'm dialing over voip to another domain. Everything swings except that when a remote party hangs up, I don't get a disconnect signal at my caller. Instead I'm getting ". I'm pretty sure I messed up something with my force/unforce proxy but I just can't get it working right. Note that when I'm calling to the PSTN (same transit proxy) through Asterisk I do get the disconnect signal. Any help would be appreciated :). Below is the config of my local proxy. I can post my transit proxy conf if needed. Thanks, Maciek debug=3 fork=no log_stderror=yes listen=<local_proxy> port=5060 children=4 mpath="/usr/local/lib/openser/modules/" dns=no rev_dns=no alias="yam" fifo="/tmp/ser_fifo" fifo_db_url="mysql://openser:openserrw@localhost:3306/openser" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "mysql.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "uri.so" loadmodule "uri_db.so" loadmodule "domain.so" loadmodule "nathelper.so" loadmodule "textops.so" loadmodule "avpops.so" loadmodule "permissions.so" loadmodule "uac.so" loadmodule "acc.so" loadmodule "auth_radius.so" loadmodule "group_radius.so" loadmodule "avp_radius.so" loadmodule "xlog.so" loadmodule "cpl-c.so" loadmodule "exec.so" #modparam("auth_db","calculate_ha1",1) #modparam("auth_db","password_column","password") modparam("auth_db|permissions|uri_db|usrloc|cpl-c","db_url","mysql://openser:openserrw@localhost:3306/openser") modparam("usrloc", "db_mode", 2) modparam("usrloc|registrar","use_domain",1) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 120) #modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("nathelper","rtpproxy_sock","unix:/var/run/rtpproxy.sock") modparam("nathelper","natping_interval",0) modparam("nathelper","ping_nated_only",1) modparam("registrar","nat_flag",6) modparam("registrar","sip_natping_flag",7) modparam("avpops", "avp_url", "mysql://openser:openserrw@localhost:3306/openser") modparam("avpops", "avp_table", "usr_preferences") modparam("avpops", "use_domain", 1) modparam("acc", "radius_flag",1) modparam("acc", "radius_missed_flag", 2) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 1) modparam("acc", "service_type", 15) modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp") modparam("acc", "report_ack",1) modparam("acc|auth_radius|group_radius|avp_radius","radius_config","/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("group_radius", "use_domain", 1) modparam("avpops", "avp_aliases", "day=i:101;time=i:102") modparam("cpl-c","cpl_table","cpl") modparam("cpl-c","cpl_dtd_file","/usr/local/etc/openser/cpl-06.dtd") #modparam("cpl-c","log_dir","/var/log/openser") modparam("cpl-c","lookup_domain","location") modparam("cpl-c","use_domain",1) modparam("cpl-c","nat_flag",6) modparam("cpl-c","proxy_recurse",2) modparam("cpl-c","proxy_route", 0) #modparam("cpl-c","timer_avp","i:5") route{ #-------------------------------------- # Sanity Check Section #-------------------------------------- if (!mf_process_maxfwd_header("10")){ sl_send_reply("483", "Too Many Hops"); return; }; if (msg:len > max_len){ sl_send_reply("513", "Message Overflow"); return; }; #-------------------------------------- # Record Route Section #-------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL"){ xlog("L_INFO","$tu\n $fu\n $ru\n"); unforce_rtp_proxy(); }; #-------------------------------------- # Loose Route Section #-------------------------------------- if (loose_route()) { log(1,"loose route\n"); if (is_method("BYE")) { unforce_rtp_proxy(); sl_send_reply("200","Bye"); # acc_rad_request("200"); # acc_log_request("200"); } if((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403","Forbidden"); return; }; if (method=="INVITE") { if (!allow_trusted()) { if (!radius_www_authorize("")) { www_challenge("","0"); return; } else if(!check_from()) { sl_send_reply("403","Use From=ID"); return; }; consume_credentials(); }; if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; # force_rtp_proxy("1"); force_rtp_proxy("s"); }; route(1); return; }; #-------------------------------------- # Call Type Processing Section #-------------------------------------- if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { sl_send_reply("403", "Forbidden"); }; return; }; if (method=="ACK") { route(1); return; } else if(method=="CANCEL"){ route(1); return; } else if (method=="INVITE"){ if(!cpl_run_script("incoming", "is_stateless")) { # script execution failed t_reply("500", "CPL script execution failed"); }; route(3); return; } else if (method=="REGISTER") { cpl_process_register(); route(2); return; }; lookup("aliases"); if (!is_uri_host_local()) { route(4); route(1); return; }; if (!lookup("location")){ # xlog("L_INFO"," $ru \n, $fu \n, $tu \n"); sl_send_reply("404", "User Not Found"); return; }; route(1); } route[1] { #-------------------------------------- # Default Message Handler #-------------------------------------- t_on_reply("1"); if (!t_relay()) { if(method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { #-------------------------------------- # REGISTER Message Handler #-------------------------------------- sl_send_reply("100", "Trying"); if(!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100","Trying"); if(!radius_www_authorize("")) { www_challenge("","0"); return; }; if(!check_to()){ sl_send_reply("401","Unauthorized"); return; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { #----------------------- # INVITE Message Handler #----------------------- if (!allow_trusted()) { if(!radius_www_authorize("")) { www_challenge("","0"); return; } else if (!check_from()){ sl_send_reply("403","Use From=ID"); return; }; consume_credentials(); }; if(nat_uac_test("19")) { setflag(6); }; avp_load_radius("caller"); # setflag(1); # setflag(2); route(9); # CLIP/CLIR route(8); # Check if number is blocked route(7); # Set/remove call forwarding route(10); # VPABX lookup("aliases"); if(!is_uri_host_local()) { route(4); route(1); return; }; if (avp_db_load("$ru/username", "$avp(s:callfwd)")) { setflag(22); avp_pushto("$ru", "$avp(s:callfwd)"); route(6); return; }; if (!lookup("location")) { #Outbound domain if (uri=~"^sip:[0-9]{10}@") { route(4); route(5); return; }; sl_send_reply("404", "User Not Found"); return; }; if (avp_db_load("$ru/username", "$avp(s:fwdonbusy)")) { if (!avp_check("$ru(s:fwdonbusy)", "eq/$ru/i")) { setflag(26); }; }; if (avp_db_load("$ru/username", "$avp(s:fwdnoanswer)")) { if (!avp_check("$avp(s:fwdnoanswer)", "eq/$ru/i")) { setflag(27); }; }; t_on_failure("1"); route(4); route(1); } route[4] { # ---------------------- # NAT Traversal Section # ---------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); # force_rtp_proxy("i"); force_rtp_proxy(); }; } route[5] { #---------------------- # Main Domain Handler #---------------------- if (uri=~"^sip:022448755[0-9]@"){ sl_send_reply("404","User Not Found"); drop; } xlog("L_INFO","Main domain Handler: address changed to <outbound_proxy>\n"); #rewritehostport("<outbound_proxy>:5061"); rewritehost("<outbound_proxy>"); t_on_failure("1"); route(1); } route[6] { # ------------------------- # Call Forwarding Handler # ------------------------- if (uri=~"^sip:1[0-9]{10}@") { strip(1); }; lookup("aliases"); revert_uri(); append_branch(); if (!is_uri_host_local()) { if (!isflagset(22)) { append_branch(); }; route(4); route(1); return; }; if (uri=~"^sip:00[0-9]{10}@") { route(4); route(1); return; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { route(4); route(1); return; }; sl_send_reply("404", "3User Not Found"); }; route(4); route(1); } route[7] { ####################################### # Call forwarding setters ####################################### if(uri=~"^sip:\*21\*"){ subst_uri('/^sip:\*21\*(.*)@(.*)$/sip:\1@yam/'); avp_db_delete("$fu","$avp(s:callfwd)"); avp_write("$ru","$avp(s:callfwd)"); avp_db_store("$fu","$avp(s:callfwd)"); sl_send_reply("603", "Call forwarding set"); drop; }; if(uri=~"^sip:\*31\*"){ subst_uri('/^sip:\*31\*(.*)@(.*)$/sip:\1@yam/'); avp_db_delete("$fu","$avp(s:fwdonbusy)"); avp_write("$ru","$avp(s:fwdonbusy)"); avp_db_store("$fu","$avp(s:fwdonbusy)"); exec_msg("/home/pgrabowski/busy_fwd_cpl"); sl_send_reply("603", "Call forwarding on busy set"); drop; }; if(uri=~"^sip:\*41\*"){ subst_uri('/^sip:\*31\*(.*)@(.*)$/sip:\1@yam/'); avp_db_delete("$fu","$avp(s:fwdnoanswer)"); avp_write("$ru","$avp(s:fwdnoanswer)"); avp_db_store("$fu","$avp(s:fwdnoanswer)"); exec_msg("/home/pgrabowski/noanswer_fwd_cpl"); sl_send_reply("603", "Call forwarding on no answer set"); drop; }; if(uri=~"^sip:\*22\*"){ avp_db_delete("$fu","$avp(s:callfwd)"); sl_send_reply("603", "Call forwarding removed"); drop; }; if(uri=~"^sip:\*32\*"){ avp_db_delete("$fu","$avp(s:fwdonbusy)"); exec_msg("/home/pgrabowski/ser_work/rm_busy_fwd_cpl"); sl_send_reply("603", "Call forwarding on busy removed"); drop; }; if(uri=~"^sip:\*42\*"){ avp_db_delete("$fu","$avp(s:fwdnoanswer)"); exec_msg("/home/pgrabowski/ser_work/rm_noanswer_fwd_cpl"); sl_send_reply("603", "Call forwarding on no answer removed"); drop; }; } route[8] { ################################## # Outgoing number blocking ################################## if(avp_check("$avp(s:caller_blocked)","eq/$tu")){ sl_send_reply("403", "Forbidden"); return; }; } route[9] { ################################# # CLIP/CLIR ################################# if(is_avp_set("$avp(s:caller_clip)")){ uac_replace_from("$avp(s:caller_clip)",""); }; if(is_avp_set("$avp(s:clir)")){ uac_replace_from("Restricted","sip:restricted@<local_proxy>"); }; } route[10] { ################################# # VPABX ################################# if(uri=~"^sip:[0-9]{3}@"){ if (is_avp_set("$avp(s:caller_vpabx)")) { xlog("L_INFO","VPABX\n"); subst_uri('/^sip:(.*)$/sip:$avp(s:caller_vpabx)\1/'); xlog("$ru $tu\n"); } } } onreply_route[1] { if (isflagset(6) && (status=~"(180)|(183)|2[0-9][0-9]")) { if(!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } failure_route[1] { if (t_check_status("487")) { return; }; subst_uri('/^sip:(.*)@(.*)$/sip:100.\1@<local_proxy>/'); rewritehostport("<local_proxy>:5061"); append_branch(); t_relay(); } ---------------------------------------------------------------------- smieszne, muzyka, pilka, sexy, kibice, kino, ciekawe, extreme, kabaret http://link.interia.pl/f19d4 - najlepsze filmy w intermecie

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