sr-users September 2006

sr-users@lists.kamailio.org

184 participants
436 discussions

by Bogdan-Andrei Iancu

Hi everybody, I just commited on the CVS a grand ACC restructuring and clean-up. Here is form commit log: - simplify internal logic for preparing and message selection - more efficient data collection - eliminate the FMT-based accounting; instead of it, a default set of basic values are all the time accounted; for more, extra accounting should be used -> this leads to a more selective / flexible / performant accounting - pre-builds sets of names / columns / RADIUS AVPs / DIAMETER AVPs instead of doing it each time a request is accounted - "n/a" string for not found/available values replaced with empty string "" - eliminated a lot of un-necessary functions - heritage from the all TM callback design (pre pre-transaction transaction) - DB accounting table redesigned - accounting logic separated in a new file for a better modularity. - implemented an accounting environment for storing data; helps in reducing the number of function's parameters and of internal functions; more important, useless conversions from str to int and to int again are eliminated The most important change was the drop of FMTs for a more flexible system: have a basic set of values used of call matching, plus the usage of extra accounting for anything more. The basic set is defined to : method name From URI From TAG To URI To TAG CallID Cseqno Reply code Reply reason Time stamp having a timestamp generated by the proxy instead by the backend (syslog, RADIUS server, SQL server) is important as it is more accurate - the delays due transport to the backend server are eliminated. Also the time is logged in UNIX timestamp format to avoid confusions about timezones. From how to use it (enabling acc via flags), nothing changed :)..only some module parameters were lost. For more details, please see: http://openser.org/docs/modules/1.2.x/acc.html any feedback is welcomed regards, bogdan

17 years, 10 months

Re: [Users] Checking for Digit Length

by Mark Kent

> Basically I am trying to find out a way of checking for minimum and > maximum digit length dialled by our users, in order to filter some > of the traffic through our OpenSER server. Dialling rules aside, is > there a way? Unless I missed it, no one told the guy what he needs to know. Instead, he is pointed at a lengthy specification for posix regular expressions. I know there is value in teaching a guy to fish, but at some point someone has to just answer the question... This if (uri=~"sip:[0-9]{10,24}") { ... } will match a sequence of digits between 10 and 24 in length (inclusive). Now, the above is not a great test in practice. I don't know how long the longest number can be outside the North American Numbering plan (where it is relatively easy: 1-NXX-NXX-XXXX, see nanpa.com). And there are restrictions on the first few digits that you might want to check, depending on where you are in the world. I'm sure that someone reading this list has already got the ultimate, global, number checker coded for openser... it's just a question of whether they'll cough it up in public. -mark

17 years, 10 months

[Serusers] Active Session Count in RTPProxy

by Gaurav Kulshreshtha

Is it possible in someway to extract the actual number of active sessions in RTP Proxy. There is sessions_created variable but its a cumulative count of sessions created. Gaurav

17 years, 10 months

[Users] compiling OpenSER on Solaris 10 x86

by Nick De Cristofaro

I'm having a lot of difficulty compiling OpenSER on Solaris 10. Firstly I was missing a lot of packages which I needed to download from sunfreeware.com (gcc, ld) the /usr/bin/ccs version of make does not work on the makefile here is the error: bash-3.00# make make: Fatal error in reader: Makefile.sources, line 18: Unexpected end of line seen I looked in the file but I don't see how it would complaing of some unexpected end of line. I then followed to try GNU's make I downloaded. and then I get an error on the /usr/bin/ccs/lex and yacc's (it mentioned to use yacc and not bison) the error was: bash-3.00# /usr/local/bin/make all yacc -d -b cfg cfg.y conflicts: 1 shift/reduce lex cfg.lex "cfg.lex":line 102: Error: missing translation value make: *** [lex.yy.c] Error 1 Then I decided to try flex instead of lex. bash-3.00# /usr/local/bin/make all yacc -d -b cfg cfg.y conflicts: 1 shift/reduce flex cfg.lex flex: fatal internal error, exec failed make: *** [lex.yy.c] Broken Pipe make: *** Deleting file `lex.yy.c' so I'm kindda stuck, it's apparently been compiled on the x68 s10 platform before so there must be some sort of makefile modifications I must make everywhere ? Anyone know what may be going on? I was hoping there would be a package OpenSER version for solaris built somewhere :S Thanks! Nick

17 years, 10 months

[Users] Local authentication

by Artur Hayne

Hello, Can I authenticate the openser locally? I want the user that do local login (in the computer of the server) can be authenticate in openser server. Its possible? Thanks. --------------------------------- Yahoo! Search Música para ver e ouvir: You're Beautiful, do James Blunt

17 years, 10 months

[Users] rtpproxy bridge mode with multiple handsets per extension

by David Carter

Good afternoon, I have a small openser test rig with rtpproxy running as a bridge between two networks: rtpproxy -l <public IP>/<private IP> -s /tmp/rtpproxy.sock This works well apart from one issue. I have set up two handsets for an extension, one on the public network and one on the private network. The idea is to provide one phone in the office and one at home, so I can pick up at either location when someone calls. If I call the extension both handsets ring correctly, but audio is only set up correctly for one of the two handsets when I pick up. Everything works fine if both phones are on the public or the private network. I infer that this is because primitives working on the ruri and friends: avp_write("$ruri/domain", "i:20"); only work on the address of the first registered handset that they find, even if lookup("location") has created multiple branches. Does anyone have any ideas or workarounds which would allow me to set up multiple handsets for a given extension when bridge mode is involved? I tried abusing the aliases table to redirect an incoming call onto multiple SIP URIs. It looks like rtpproxy indexes by Call-ID, and hence struggles with multiple branches which share a single Call-ID. Thanks, -- David Carter Email: David.Carter(a)ucs.cam.ac.uk University Computing Service, Phone: (01223) 334502 New Museums Site, Pembroke Street, Fax: (01223) 334679 Cambridge UK. CB2 3QH.

17 years, 10 months

[Serusers] Re: Re: Using IP address as domain

by mukesh

Hi andrey, I have made suggested changes in ser.cfg as follows listen=202.87.x.x alias=202.87.x.x and did #export SIP_DOMAIN=202.87.x.x I get following log when try to start SER #/etc/rc.id/init.d/ser start ------------log starts here-------------------- Starting ser: 0(5116) WARNING: fix_socket_list: could not rev. resolve 202.87.x.x 0(5116) WARNING: fix_socket_list: could not rev. resolve 202.87.x.x WARNING: no fork mode stateless - initializing 0(0) Maxfwd module- initializing textops - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 Listening on udp: 202.87.x.x [202.87.x.x]:5060 tcp: 202.87.x.x [202.87.x.x]:5060 Aliases: *: 202.87.x.x:* 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 0(0) ERROR: udp_init: bind(6, 0x810237c, 16) on 202.87.x.x: Cannot assign requested address [FAILED] ------------log ends here---------------------- please advice. Regards Mukesh From: "Andrey Kuprianov" <andrey.kouprianov(a)gmail.com> Subject: Re: [Serusers] Using IP address as domain To: serusers(a)iptel.org Message-ID: <ab5f5bda0609210935y71edb46ci6ea9966b170e52de(a)mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi, AFAIK, SIP_DOMAIN you need to export anyway.. Just put your IP instead of the name. Then this SIP_DOMAIN will be used to properly add contacts to your subscriber table. So, if your domain is an IP address, newly added ppl will have an IP instead of domain for their host part of URI. Try also to specify alias as you IP address, i.e. listen="202.x.x.x" alias="202.x.x.x" You may also try to add a domain entry to the domain table with serctl (that is if you use functions in your config such as "is_uri_host_local()"), i.e. serctl domain add 202.x.x.x Hope it will work for you. Andrey. On 9/20/06, mukesh <mukesh(a)hungama.com> wrote: > Hi friends, > I have SER (with database support) running on my linux box. But I do > not have a domain name registered for the IP of this machine but it does > have a public IP (202.87.xx.xxx) address which allows me access it from > outside. > > If I try to access (using sipsak client) a uri from local system it > fails. > example URIs I am trying are ... > sip:mukesh@<public ip> > sip:mukesh@<public ip>:5060 > > ...with no result :( > > I referred http://mit.edu/sip/sip.edu/ser.shtml to configure my > ser.cfg. > > So the question is How can I connect to SER using IP and do I need > to use "# export SIP_DOMAIN=<public ip>"? > > Regards > Mukesh

17 years, 10 months

[Serusers] Re: Re: Using IP address as domain

by mukesh

Hi andrey, I have made follwing changes in ser.cfg listen="202.87.x.x" alias="202.87.x.x" #export SIP_DOMAIN = 202.87.x.x when I run # /etc/rc.d/init.d/ser restart I get this log ----------------------log starts here ------------------------- Starting ser: 0(5116) WARNING: fix_socket_list: could not rev. resolve 202.87.x.x 0(5116) WARNING: fix_socket_list: could not rev. resolve 202.87.x.x WARNING: no fork mode stateless - initializing 0(0) Maxfwd module- initializing textops - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 Listening on udp: 202.87.x.x [202.87.x.x]:5060 tcp: 202.87.x.x [202.87.x.x]:5060 Aliases: *: 202.87.x.x:* 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 0(0) ERROR: udp_init: bind(6, 0x810237c, 16) on 202.87.x.x: Cannot assign requested address [FAILED] [root@oradb1 /]# ----------------------log ends here ------------------------- Regards Mukesh From: "Andrey Kuprianov" <andrey.kouprianov(a)gmail.com> Subject: Re: [Serusers] Using IP address as domain To: serusers(a)iptel.org Message-ID: <ab5f5bda0609210935y71edb46ci6ea9966b170e52de(a)mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi, AFAIK, SIP_DOMAIN you need to export anyway.. Just put your IP instead of the name. Then this SIP_DOMAIN will be used to properly add contacts to your subscriber table. So, if your domain is an IP address, newly added ppl will have an IP instead of domain for their host part of URI. Try also to specify alias as you IP address, i.e. listen="202.x.x.x" alias="202.x.x.x" You may also try to add a domain entry to the domain table with serctl (that is if you use functions in your config such as "is_uri_host_local()"), i.e. serctl domain add 202.x.x.x Hope it will work for you. Andrey. On 9/20/06, mukesh <mukesh(a)hungama.com> wrote: > Hi friends, > I have SER (with database support) running on my linux box. But I do > not have a domain name registered for the IP of this machine but it does > have a public IP (202.87.xx.xxx) address which allows me access it from > outside. > > If I try to access (using sipsak client) a uri from local system it > fails. > example URIs I am trying are ... > sip:mukesh@<public ip> > sip:mukesh@<public ip>:5060 > > ...with no result :( > > I referred http://mit.edu/sip/sip.edu/ser.shtml to configure my > ser.cfg. > > So the question is How can I connect to SER using IP and do I need > to use "# export SIP_DOMAIN=<public ip>"? > > Regards > Mukesh

17 years, 10 months

RE: [Serusers] SIPP with SER

by Verbeiren, David

> Hello , > > has anyone any experience on testing SER with SIPP ?? > > I've sent the following mail to sipp-users mailing list, I'm sending it > also here, in case anyone has any idea or experience. This may not be exactly what you want to test but I've done tests with SIPp and various SIP proxies, SER being one of them, by targeting the UAS itself on the SIPp command line but with a "remote sending address" parameter specifying the proxy to use. This works great and does not require the 'service' user to be known to SER as it simply looks at the hostname (or ip address) part of the URI to proxy the call. This gives the following modified command lines: UAC : ./sipp -sn uac 2.2.2.2 -r 1 -m 1 -d 5000 -i 1.1.1.1 -rsa 3.3.3.3 UAS : ./sipp -sn uas -rsa 3.3.3.3 The remote sending address is similar to the "outbound proxy" parameter found on many SIP phones. This way of running SIPp gives the call flow you describe but shortcuts most of the ser.cfg routing logic as uri!=myself. Regards, -David

17 years, 10 months

[Serusers] problem connecting two asterisk servers through SER

by Sameer Ahmed

Hi We have two asterisks registered as UA's. SER is acting as a proxy server. We want to make calls between two asterisk users through SER. One asterisk server has extensions starting with 2, and the other one has extensions starting with 4. We route the calls based on the first digit. We are unable to call from an extension on one asterisk server to the other asterisk server. "rewritehostport" doesn't seem to be doing the trick for us. Does anyone have a solution? Below is our ser.cfg file: =====================SER.CFG===================== # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=9 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines for debugging */ #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_mode=0666 fifo_db_url="mysql://ser:heslo@192.168.1.60 /ser" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #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; }; if(method == "INVITE") { if (uri =~ "sip:2[0-9]{3}@*"){ log(1, "Forwarding to PakVista BlueFlash\n"); route(2); break; } else if (uri =~ "sip:4[0-9]{3}@*"){ log(1, "Forwarding to MyRoom BlueFlash\n"); route(3); break; } } log(1, "After INVITE\n"); # 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(); log(1, "After REGISTER\n"); # 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) { log(1,"Inside NOT uri==myself \n"); # 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) { log(1, "Inside uri==myself\n"); if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("192.168.1.60", "subscriber")) { www_challenge(" 192.168.1.60", "0"); break; }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { log(1, "location Not Found\n"); sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { log(1, "Inside route 1\n"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } route[2] { log(1, "Inside route 2\n"); rewritehostport("192.168.1.61:5060"); if (!t_relay()) { log(1, "t_relay error\n"); sl_reply_error(); }; log(1, "Leaving route 2\n"); } route[3] { log(1, "Inside route 3\n"); rewritehostport("192.168.1.62"); if (!t_relay()) { log(1, "t_relay error\n"); sl_reply_error(); }; #forward(uri:host, uri:port); log(1, "Leaving route 3\n"); ===============================EOF========================

17 years, 10 months

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sr-users September 2006