sr-users January 2005

sr-users@lists.kamailio.org

199 participants
399 discussions

[Serusers] SER and NAT and RTPproxy
by Ashling O'Driscoll 30 Jan '05

30 Jan '05

Hi, Thank you to everyone who has replied to me to date. Unfortunately the problem is still there. Regarding Mr B's reply below. The BT100 client is behind NAT and I have rtp ports opened on my linksys router with port forwarding enabled. I have recaped the problem again below and would really appreciate any more thoughts or commenst to help resolve this problem as soon as possible...My demo this week will not go well if I cant figure this out!! Recap: Basically I think the problem comes down to nat. I still get the rtp errors mentioned in below emails in the var/log/messages file whether voice works or not(strangely enough)...but here are the scenarios that work and dont work: I have ser and asterisk on a private natted network. They are reached via a router which has a public address and does port forwarding. One my clients(a bt100 hardphone is also on this lan but registers with ser throught the public address anyway...so my understanding is that the bt100's sip register message goes out onto the internet and back in again. Now if the bt100 rings a client (e.g. xlite or windows messenger)or vice versa that are also on this lan...the call works and voice is transmitted (even though they again register through the public address by routing their sip messages onto the internet and back in again). However if the bt100 tries to ring a client not on the lan (on another network which may or may not be behind nat)...np voice is transmitted....Surely this is a nat problem even through the rtp errors are being displayed??.... If anybody has any idea how to fix this or if any more information is required in order to troubleshoot please let me know. Like I mentioned before this must be demoed next week...This natting scenarion worked when clients registered direct to asterisk so i presume its possible with ser. Thanks in advance, Aisling. From: "Mr B" <"cabalitomb atshaw.ca"@fox.iptel.org> Subject: RE: Re: [Serusers] SER and NAT and RTProxy To: serusers(a)iptel.org Message-ID: <0IB3004283LBCJ@l-daemon> Content-Type: text/plain; charset=us-ascii What I have found on the BT-100 (or any of the BT Series) is that the unit does not function very well behind any NAT without a RTP port open. Also you may want to check your router if you are using one. These are my findings of routers SMC - seems to work the best for VOIP Linksys - okay D-Link - very problematic - seems to always retain some old info Just my .000001% of a cent opinion Peter -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@iptel.org] On Behalf Of Ashling O'Driscoll Sent: Saturday, January 29, 2005 8:56 AM To: serusers(a)iptel.org Subject: FW: Re: [Serusers] SER and NAT and RTProxy After doing a small bit of testing I have a bit more information which may help identify the problem....Basically I think the problem comes down to nat. I still get the rtp errors mentioned below in the var/log/messages file whether voice works or not(strangely enough)...but here are the scenarios that work and dont work: I have ser and asterisk on a private natted network. They are reached via a router which has a public address and does port forwarding. One my clients(a bt100 hardphone is also on this lan but registers with ser throught the public address anyway...so my understanding is that the bt100's sip register message goes out onto the internet and back in again. Now if the bt100 rings a client (e.g. xlite or windows messenger)or vice versa that are also on this lan...the call works and voice is transmitted (even though they again register through the public address by routing their sip messages onto the internet and back in again). However if the bt100 tries to ring a client not on the lan (on another network which may or may not be behind nat)...np voice is transmitted....Surely this is a nat problem even through the rtp errors are being displayed??.... If anybody has any idea how to fix this or if any more information is required in order to troubleshoot please let me know. Like I mentioned before this must be demoed next week...This natting scenarion worked when clients registered direct to asterisk so i presume its possible with ser. Thanks in advance, Aisling. ---- Original Message ---- From: ashling.odriscoll(a)cit.ie To: serusers(a)iptel.org Subject: FW: Re: [Serusers] SER and NAT and RTProxy Date: Sat, 29 Jan 2005 13:58:14 -0000 Unfortunately this still hasnt worked. I changed the modparam line in ser.cfg and ran rtpproxy again as follows: cd /root/Desktop/rtpproxy ./rtpproxy -s 127.0.0.1 Then I restarted SER. However now voice isnt being transmitted AT ALL...and i cant make it work even by running: cvs -d:pserver:anonymous@cvs.ser.berlios.de:/cvsrot/ser co rtpproxy Whats so weird is that I tested all this on wed evening and everything worked fine, now I repeatedly get the rtp errors documented in my first email on the /var/log/messages file. Does anyone have any more ideas...Im supposed to be demonstrating this sytem as prototype next week and it wont look very impressive if voice wont transmit.... Very stumped, Aisling. ---- Original Message ---- From: info(a)marikar.com To: serusers(a)iptel.org Subject: Re: [Serusers] SER and NAT and RTProxy Date: Sat, 29 Jan 2005 02:30:43 +0100 >-----BEGIN PGP SIGNED MESSAGE----- >Hash: SHA1 > >Hello, > >I got the same error-messages. The rtpproxy works fine if I change >the line >modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock") >to modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1") and start >the >rtpproxy with the option -s 127.0.0.1 > >I hope this will help, > >Achim > >Am Donnerstag, 27. Januar 2005 22:01 schrieb Ashling O'Driscoll: >> Hi all, >> >> I have a strange problem with the audio with some calls. I have >setup >> RTPProxy and nathelper modules. I came across an error saying the >RTP >> proxy was disabled but solved it by searching the archives and >> executing: >> >> cvs -d:pserver:anonymous@cvs.ser.berlios.de:/cvsroot/ser co >rtpproxy. >> >> This worked fine and my audio was transmitted. However every now >and >> again (apparently at random) my audio doesnt work.When I look at >the >> error logs in /var/log/messages, I see the following: >> >> ERROR: send rtpp_command: cant read reply from a rtp proxy >> WARNING: rtpp_test: cant get version of the RTP proxy >> WARNING: rtpp_test: support for the rtp proxy has been temporarily >> disabled >> ERROR: force_rtp_proxy2: support for porxy disabled. >> >> The CVS command fixes it temporarily. Does this mean I just have to >> run the command randonly every so often?...Is there a way to >> permanently fix this? >> >> Thanks, >> Aisling. >> -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

1 0

[Serusers] SER and serctl problem
by arda balkanay 30 Jan '05

30 Jan '05

Hi i've installed SER 0.8.12 with mysql-support from the rpm file in RH9 environment. bur when i tried to use it in the first time i gor some error messages like [root@ardaRH9 RPMS]# /etc/rc.d/init.d/ser start Starting ser: [ OK ] [root@ardaRH9 RPMS]# /etc/rc.d/init.d/ser stop Stopping ser: [FAILED] [root@ardaRH9 RPMS]# serctl moni Error opening ser's FIFO /tmp/ser_fifo Make sure you have line fifo=/tmp/ser_fifo in your config [root@ardaRH9 RPMS]# /etc/rc.d/init.d/ser restart Stopping ser: [FAILED] Starting ser: [ OK ] [root@ardaRH9 RPMS]# serctl moni Error opening ser's FIFO /tmp/ser_fifo Make sure you have line fifo=/tmp/ser_fifo in your config when i looked at /tmp i couldn't see ser_fifo file what can be the reason an d how can i solve the problem thanks best regards Arda Balkanay

1 0

[Serusers] HowToForAll MySQL Accounting PSTN update
by mr. barker 30 Jan '05

30 Jan '05

Update - the gmake line from gmake all mode=debug exclude_modules="" modules To gmake all bin mode=debug exclude_modules="" modules this will make a file called ser-0.8.14_linux_i386.tar.gz this will be located in the very root of the drive I believe if not just do a search for the file. Any response to the success or failure would be apprectiated. Thanks Peter

1 0

[Serusers] HowToForAll MySQL Accounting PSTN
by mr barker 30 Jan '05

30 Jan '05

hope that this is not to large of a message or that it doesnt get bounced my emails do not seem to be going through. This has taken me quite some time to do this as I come from the Windows world. Getting into the Linux world was quite a jump. (Now the rant) First I would like to say that if some of the people in the Linux world had better documentation and most importantly keeping it current with the releases of software it would cut down on development time. I use Konqueror for getting around the Linux System like using windows file explorer except better. Open up a Konqueror and press F9 that will open a side bar of file management. F4 is to open up a command shell in the directory that you are in. This is based on the compiled from src Stable Version Of SER 8.14 Located Here ftp.berlios.de/pub/ser/0.8.14/src/ put this in the very root of the hard drive (this should keep all the paths correct dont change the name of the directory from sip_router) I am running Fedora Core 3 Distribution Located here http://download.fedora.redhat.com/pub/fedora/linux/core/3/i386/iso/ there are 4 isos (FC3-i386-disc1-4.iso) which are bootable. Make sure that MySQL is enabled (make sure you check it off when installing) along with PhPMySQL I installed the Server Packages from the list. I also installed Webmin located here http://www.webmin.com/ for remote access via a M$ machine. Compiling the SRC code was a bit problematic for me (Note that there is an Install Doc error when making the bin distro you will need to go into the MakeFile and comment out the line at the bottom) it is only documentation. You will see this as soon as you use the gmake utility. You will first need to edit the Makefile in the module of ACC locate this line # uncomment the next line if you wish to enable SQL accounting DEFS+=-DSQL_ACC This is the way it should look uncommented. Now save it. Lol Open up a command shell in the sip_router directory. Type the following gmake all mode=debug exclude_modules="" modules you will get some errors that are caused by the imcompleteness but not to worry you wont be using them .. It will should come up with ser-0.8.14_linux_i386.tar.gz you will need to do a find for this file because I have no idea where it places it so I use Konqueror to locate it. Then do a copy command to the Home directory so you have it for later also. You will need to do some reading in the manual of SER so you will be familiar with the system. But here is a working ser.cfg using MySQL for accounting You will need to create the data base using the shell script in the SBIN directory. If you use serctl start and the pid file is not found then that means that there is an error in the ser.cfg Yes I know that the code looks ugly however so does your dog ... just kidding it is a work in progress as I work it down more I hope that this will help a few folks out. # This is a reworked script from someone else. # $Id: ser.cfg A working MySQL, Accounting PSTN Reworked for 8.14 # By Peter Bjorklund / Mr. B .. may the code gods be pleased .. lol # # You will need the adjust the ports to suit you needs # # xxx.xxx.xxx.xxx this is your IP address - ex. 123.456.789.123 # yyy.yyy.yyy.yyy this is your PSTN GateWay address ex. 123.456.789.123 # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) memlog=3 listen=xxx.xxx.xxx.xxx # this is your IP address sip_warning=yes /* Uncomment these lines to enter debugging mode debug=8 fork=no log_stderror=yes */ check_via=yes # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- 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/acc.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/exec.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/print.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri.so" # ----------------- setting module-specific parameters --------------- # -- tm params -- modparam("tm", "fr_timer", 10 ) modparam("tm", "fr_inv_timer", 24 ) modparam("tm", "wt_timer", 10 ) # modparam("tm", "uac_from", "sip:daemon@iptel.org" ) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # * 0 -- dont use mysql, 1 -- write_through, 2--write_back */ modparam("usrloc", "db_mode", 2) modparam("usrloc", "timer_interval", 10) modparam("usrloc", "db_url","mysql://ser:heslo@localhost/ser") modparam("group", "db_url","mysql://ser:heslo@localhost/ser") # -- auth params -- modparam("auth_db", "db_url","mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") # -- acc params -- # report ACKs too for sake of completeness -- as we account PSTN # destinations which are RR, ACKs should show up modparam("acc", "db_url","mysql://ser:heslo@localhost/ser") # modparam("acc", "log_missed_flag", 3) modparam("acc", "log_level", 1) modparam("acc", "db_flag", 1) modparam("acc", "report_ack", 0) modparam("acc", "log_fmt", "miocfsu") ############################################ # ------------------------- request routing logic ------------------- # main routing logic route{ /* ********* ROUTINE CHECKS ********************************** */ # filter too old messages if (!mf_process_maxfwd_header("10")) { log("LOG: Too many hops\n"); sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; ############# deal with local area code ################# you can replace the xxx with your area code if (uri=~"sip:xxx[2-9][0-9][0-9][0-9][0-9][0-9][0-9]@.*") { strip(3); # search_append("From: .*<sip:", "xxx"); # if (search("^To: .*<sip:xxx")) { # replace("To: .*<sip:xxx", "To: <sip:"); # }; }; ################# billing ########################### if (method=="INVITE" || method=="BYE") { setflag(1); setflag(2); # yes I use this flag for debuging }; /* ********* RR ********************************** */ /* Do strict routing if route headers present */ #if (loose_route()) { t_relay(); break; }; # do not use this format # instead just use this loose_route(); /* ********* DIVERSION ********************************** */ /* apply all diversions before we proceed with processing of requests for us */ ########## You can adjust the Diviersions to your needs ################### /* IM gateway diversions */ #if (uri=~"sip:.*@icq\yourdomainname\.com" # | uri=~"sip:.*@msn\.yourdomainname\.com" # | uri=~"sip:.*@aim\.yourdomainname\.com" # | uri=~"sip:.*@yahoo\.yourdomainname\.com" ) { # append_hf("P-hint: IMGW\r\n"); # if (!t_relay("xxx.xxx.xxx.xxx", "5070")) { # sl_reply_error(); # }; # break; # }; /* divert voicemail requests */ #if (uri=~"mail\.yourdomainname\.com" | uri=~":5066"| uri=~":6060") { # sethost("yourdomainname.com"); # append_hf("P-hint: VOICEMAIL\r\n"); # if ( !t_relay("mail.iptel.org", "6060")) { # sl_reply_error(); # }; # break; # }; /* ********* RR ********************************** */ # look at whether we need record-routing; # - we need it for calls from gateways (otherwise, subsequent # requests from the other # party will attempt to contact gateway # directly through blocked ports) # - we need it for Windows Messanger's IM sessions to cross # some firewalls -- we force all MESSAGEs to go via our server # to avoid blocking port numbers (some firewalls can do # standard SIP but are puzzled by Microsoft's use of obsoleted # IM session model) # - some other places may decide to set the record-routing # flag (2 chosen) too; particularly, INVITEs to our gw if ( (src_ip==yyy.yyy.yyy.yyy & method=="INVITE") || method=="MESSAGE" || method=="INFO") { setflag(2); }; /* ********* check for requests targeted out of our domain... ******* */ # sign of our domain: there is @ (username), : # (nothing) or . (host) in front of our domain name ; # if none of these cases matches, proceed with proessing of # outbound requests in route[2] if (!(uri=~"[@:\.]yourdomainname\.com([;:].*)*" # ... some phones put IP address in URI instead ... | uri=~"[@:\.]xxx\.xxx\.xxx\.xxx([;:].*)*" # ... and we serve our gateway too (we RR requests to it, so that # its address may show up in subsequent requests after # rewriteFromRoute | uri=~"@xxx\.xxx\.xxx\.xxx([;:].*)*" )) { route(2); break; }; /* ************ requests for our domain ********** */ /* now, the request is for sure for our domain */ # registers always MUST be authenticated to # avoid stealing incoming calls if (method=="REGISTER") { # Make sure that user's dont register infinite loops # (note: does not match with folded lines) if (search("^(Contact|m): .*(a)(xxx\.xxx\.xxx\.xxx|yourdomainname\.com)")) { log(1, "LOG: alert: someone trying to set aor==contact\n"); sl_send_reply("476", "No Server Address in Contacts Allowed" ); break; }; # prohibit attempts to grab someone else's To address # using valid grp; the only exception is the user # 'replciator' permitted to generate 3-rd party registrations if (!www_authorize("yourdomainname.com" /* realm */, "subscriber" /* table name */ )) { # challenge if none or invalid grp www_challenge( "yourdomainname.com" /* realm */, "0" /* no qop -- some phones can't deal with it */); break; }; if (!is_user("replicator") & !check_to()) { log("LOG: To Cheating attempt\n"); sl_send_reply("403", "That is ugly -- use To=id next time"); break; }; # it is an authenticated request, update Contact database now if (!save("location")) { sl_reply_error(); }; /* XXX not tested yet t_replicate("bat.iptel.org", "5060"); */ break; }; ###############deny calls without autorization################# if (!(src_ip==xxx.xxx.xxx.xxx | src_ip==yyy.yyy.yyy.yyy| method==ACK | method=="CANCEL" )) { if (!proxy_authorize( "yourdomainname.com" /* realm */, "subscriber" /* table name */)) { proxy_challenge( "yourdomainname.com" /* realm */, "0" /* no qop */ ); break; # let's check from=id ... avoids accounting confusion } else if (method=="INVITE" & !check_from()) { log("LOG: From Cheating attempt\n"); sl_send_reply("403", "That is ugly -- use From=id next time (gw)"); break; }; }; /* some UACs might be fooled by Contacts our UACs generate to make MSN happy (web-im, e.g.) -- tell its urneachable */ if (uri=~"sip:daemon@" ) { sl_send_reply("410", "daemon is gone"); break; }; # various aliases (might use a database in future) lookup("aliases"); # check again, if it is still for our domain after aliases # we have to include '.iptel.org' not to proceed to outbound # authentication for calls to mail.iptel.org and other hosts # served by us if ( !(uri=~"[@:\.]yourdomainname\.com([;:].*)*" | uri=~"[@:\.]xxx\.xxx\.xxx\.xxx([;:].*)*" )) { route(5); break; }; # now check if it's about PSTN destinations through our gateway; # note that 1.... and 0.... is exempted for numerical non-gw destinations if (uri=~"sip:\+?[0-1][1-9][0-9]*@.*") { route(3); break; }; # if relay if (uri=~"sip:\+?[2-9][0-9][0-9][2-9][0-9][0-9][0-9][0-9][0-9][0-9]@.*") { route(6); break; }; # does the user wish redirection on no availability? (i.e., is he # in the voicemail group?) -- determine it now and store it in # flag 4, before we rewrite the flag using UsrLoc if (is_user_in("Request-Uri", "voicemail")) { setflag(4); } ; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { # handle user which was not found ... route(4); break; }; # check whether some inventive user has uploaded gateway # contacts to UsrLoc to bypass our authorization logic if (uri=~"@yyy\.yyy\.yyy\.yyy([;:].*)*" ) { log(1, "LOG: Weird! Gateway address in UsrLoc!\n"); route(3); break; }; # if user is on-line and is in voicemail group, enable redirection # if (method=="INVITE" && isflagset(4)) { # t_on_failure("1"); # }; /* ... and also report on missed calls ... note that reporting on missed calls is mutually exclusive with silent C timer */ # setflag(3); # add RR to messages which were previously labeled for that if (isflagset(2)) { loose_route(); }; # we now know we may, we know where, let it go out now! append_hf("P-hint: USRLOC\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } #------------------- OUTBOUND ---------------------------------------- # routing logic for outbound requests targeted out of our domain # (beware, messages to our users can end up here too: for example, # an INVITE may be UsrLoc-ed, then the other party uses outbound # proxy with r-uri=the usr_loced addredd (typically IP)) route[2] { # outbound requests are allowed only for our users -- we don't # support relaying and don't like strangers bothering us # with resolving DNS; except our gateway if (!(src_ip==yyy.yyy.yyy.yyy) & !(proxy_authorize( "yourdomainname.com" /* realm */, "subscriber" /* table name */ ))) { # ACK/CANCEL have inherently no security -- just log if # included grp are wrong and proceed if (method=="BYE" ) { log("LOG: failed outbound authentication for BYE granted\n"); } else if (method=="ACK" ) { log("LOG: failed outbound authentication for ACK granted\n"); } else if (method=="CANCEL") { log("LOG: failed outbound authentication for CANCEL granted\n"); } else { proxy_challenge("yourdomainname.com" /* realm */, "0" /* no-qop */); break; }; }; # to maintain credibility of our proxy, we check From in INVITEs to be # equal to credential id so that user john.doe does not put bill.gates # in his From; we don't do that for other requests: within a dialogue, # subsequent transactions coming from the other side will have in From # To of the original transaction to match the dialog; it may be # however different from user's default From and digest; example: # user_a sends INVITE to secretary; secretary is translated to user_b; # user_b accepts and later BYEs; to match the dialog, it puts # secretary in From and sends user_b's grp -- check_from # would fail if (!src_ip==yyy.yyy.yyy.yyy& method=="INVITE" & !check_from()) { log("LOG: From Cheating attempt\n"); sl_send_reply("403", "That is ugly -- use From=id next time (OB)"); break; }; if (isflagset(2)) { record_route(); ; }; append_hf("P-hint: OUTBOUND\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } #------- ALIASED OUTBOUND -------------------------------------------- # routing logic for inbound requests aliased outbound; unlike # with real outbound requests we do not force authentication # as these calls are server by our server and we do not want # to disqualify unathenticated request originatiors from other # domains route[5] { append_hf("P-hint: ALIASED-OUTBOUND\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } #----------------- PSTN ---------------------------------------------- # logic for calls to the PSTN route[3] { # if it is a MESSAGE pass it "as is" over to our SMS gateway # (which unfortunately lives at a different host due to # lack of serial interfaces) if (method=="MESSAGE") { # set accounting setflag(1); rewritehostport("yyy.yyy.yyy.yyy:5070"); #change to our uri port for this append_hf("P-hint: SMS\r\n"); if (!t_relay()) { sl_reply_error(); }; break; }; # continue with requests to PSTN gateway ... # the international + prefix if (uri=~"sip:\+" ) { #strip(1); prefix("000"); }; # free call destinations ... no authentication needed if ( is_user_in("credentials", "free-pstn") /* free destinations */ | uri=~"sip:0@.*") { /* local PBX */ # just log it, no authentication setflag(1); setflag(2); } else { # all other PSTN destinations only for authenticated users # (Cisco GW, which has no digest support, is authenticated # by its IP address -- that's for sure not very strong; # wth confirmed that we filter packets coming from outside # and bearing SRC IP address of our network) # we are forgiving about ACK/CANCEL as digest provides no # real security for them if (!(src_ip==yyy.yyy.yyy.yyy| method==ACK | method=="CANCEL" )) { if (!proxy_authorize("yourdomainname.com" /* realm */, "subscriber" /* table name */)) { proxy_challenge("yourdomainname.com" /* realm */, "0" /* no qop */ ); break; # let's check from=id ... avoids accounting confusion } else if (method=="INVITE" & !check_from()) { log("LOG: From Cheating attempt\n"); sl_send_reply("403", "That is ugly -- use From=id next time (gw)"); break; }; }; # authorize only for INVITEs -- RR/Contact may result in weird # things showing up in d-uri that would break our logic; our # major concern is INVITE which causes PSTN costs anyway if (method=="INVITE") { # does the authenticated user have a permission for local # calls? (i.e., is he in the "local" group?) Replace xxx with your area code if (uri=~"sip:xxx[2-9][0-9]+@.*") { if (!is_user_in("credentials","local")) { sl_send_reply("403", "Local Toodle Noodle..."); break; }; # the same for long-distance } else if (uri=~"sip:1[2-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9]@.*") { #Prevent 1 900 calls etc #if (uri=~"sip:001[089]" | uri=~"sip:00900.*" ) { # sl_send_reply("403", "Added Value Destinations not permitted..."); # break; #}; if (!is_user_in("credentials","ld")) { sl_send_reply("403", "LD Toodle Noodle..."); break; }; # the same for international calls } else if (uri=~"sip:011[1-9][0-9]+@.*") { if (!is_user_in("credentials","int")) { sl_send_reply("403", "International Toodle Noodle..."); break; }; # everything else (e.g., interplanetary calls) is denied } else { sl_send_reply("403", "interplanetary Toodle Noodle..."); break; }; }; # INVITE to authorized PSTN # we passed all authorization checks for PSTN -- move on! # tag this transaction for accounting setflag(1); setflag(2); }; # authorized PSTN # requests to gateway must be record-routed because the GW accepts # only reqeusts coming from our proxy # if (isflagset(2) || method=="INVITE") if ((method=="INVITE") || (method=="ACK") || (method=="BYE")) setflag(1); # setflag(2); record_route(); # if you have passed through all the checks, let your call go to GW! # if (search('^(From|m): ".*" <sip:[2-9][0-9][0-9][0-9][0-9][0-9][0-9]@.*')) { # search_append('From: ".*" <sip:', "204"); # }; #voiplist gateway toll free only if ( uri=~"sip:1800[2-9][0-9]+@.*" | uri=~"sip:1888[2-9][0-9]+@.*" | uri=~"sip:1877[2-9][0-9]+@.*" | uri=~"sip:1866[2-9][0-9]+@.*" ) { # prefix("123456"); # replace number if your gateway uses a prefix number rewritehostport("yyy.yyy.yyy.yyy:5060"); } #voiplist LD else if ( uri=~"sip:1[1-9][0-9]*@.*" ) { # prefix("123456"); rewritehostport("yyy.yyy.yyy.yyy:5060"); } #voiplist INTERNATIONAL else if ( uri=~"sip:011[0-9][0-9]*@.*" ) { # prefix("123456"); rewritehostport("yyy.yyy.yyy.yyy:5060"); } else { sl_send_reply("403", "interplanetary Toodle Noodle..."); break; }; append_hf("P-hint: GATEWAY\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } /* *********** handling of unavailable user ******************* */ /* handling of users who are off-line */ route[4] { # user not found -- act as stateful UAS to avoid reporting # on each INVITE retranmission if (method=="INVITE" || method=="ACK" || method=="BYE" || method=="CANCEL" ) { /* requests to voicemail users will be fwded to voicemail */ if (isflagset(4)) { rewritehostport("mail.iptel.org:6060"); append_hf("P-hint: OFFLINE-VOICEMAIL\r\n"); if (!t_relay()) { sl_reply_error(); }; } else { /* non-voicemail users get 404 */ if (t_newtran()) { if (method=="ACK") { log("oops -- ACK to a non-existent transaction"); drop; }; # we reply statefuly to avoid accounting of all # retransmissions if (!t_reply("404", "Not Found")) { sl_reply_error(); }; } else { sl_reply_error(); }; }; /* we account missed calls for all off-line users */ # if (method=="INVITE") acc_request("404 Not Found"); # break; }; # non-VoIP messages: just return 404 statelessly sl_send_reply("404", "Not Found"); } ####################################### #----------------- VOIPLD --------------- # logic for calls to the VOIPLD route[6] { # if it is a MESSAGE pass it "as is" over to our SMS gateway if (method=="MESSAGE") { # set accounting setflag(1); rewritehostport("yyy.yyy.yyy.yyy:5060"); append_hf("P-hint: SMS\r\n"); if (!t_relay()) { sl_reply_error(); }; break; }; # continue with requests to PSTN gateway ... # the international + prefix if (uri=~"sip:\+" ) { #strip(1); prefix("000"); }; # free call destinations ... no authentication needed if ( is_user_in("credentials", "free-pstn") /* free destinations */ | uri=~"sip:0@.*") { /* local PBX */ # just log it, no authentication setflag(1); } else { if (!(src_ip==yyy.yyy.yyy.yyy| method==ACK | method=="CANCEL" )) { if (!proxy_authorize( "yourdomainname.com" /* realm */, "subscriber" /* table name */)) { proxy_challenge( "yourdomainname.com" /* realm */, "0" /* no qop */ ); break; } else if (method=="INVITE" & !check_from()) { log("LOG: From Cheating attempt\n"); sl_send_reply("403", "That is ugly -- use From=id next time (gw)"); break; }; }; setflag(1); }; # authorized PSTN if (isflagset(2) || method=="INVITE") record_route(); rewritehost("yyy.yyy.yyy.yyy:5060"); append_hf("P-hint: GATEWAY\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } _________________________________________________________________ Designer Mail isn't just fun to send, it's fun to receive. Use special stationery, fonts and colors. http://join.msn.com/?pgmarket=en-ca&page=byoa/prem&xAPID=1994&DI=1034&SU=ht… Start enjoying all the benefits of MSN® Premium right now and get the first two months FREE*.

1 0

[Serusers] Email not working
by Mr B 30 Jan '05

30 Jan '05

Sorry if this is a post .. admin can delete it .. my emails do not seem to be going through and I don't know why so I changed some settings now hopefully they will.

1 0

[Serusers] MySQL, Accounting, PSTN Howto for all
by mr barker 30 Jan '05

30 Jan '05

I have tried to send this via regular mail but it is not getting through ... so hence the hotmail.. hope that no one minds - Mr. B - This has taken me quite some time to do this as I come from the Windows world. Getting into the Linux world was quite a jump. (Now the rant) First I would like to say that if some of the people in the Linux world had better documentation and most importantly keeping it current with the releases of software it would cut down on development time. I use Konqueror for getting around the Linux System like using windows file explorer except better. Open up a Konqueror and press F9 that will open a side bar of file management. F4 is to open up a command shell in the directory that you are in. This is based on the compiled from src Stable Version Of SER 8.14 Located Here ftp://ftp.berlios.de/pub/ser/0.8.14/src/ put this in the very root of the hard drive (this should keep all the paths correct dont change the name of the directory from sip_router) I am running Fedora Core 3 Distribution Located here http://download.fedora.redhat.com/pub/fedora/linux/core/3/i386/iso/ there are 4 isos (FC3-i386-disc1-4.iso) which are bootable. Make sure that MySQL is enabled (make sure you check it off when installing) along with PhPMySQL I installed the Server Packages from the list. I also installed Webmin located here http://www.webmin.com/ for remote access via a M$ machine. Compiling the SRC code was a bit problematic for me (Note that there is an Install Doc error when making the bin distro you will need to go into the MakeFile and comment out the line at the bottom) it is only documentation. You will see this as soon as you use the gmake utility. You will first need to edit the Makefile in the module of ACC locate this line # uncomment the next line if you wish to enable SQL accounting DEFS+=-DSQL_ACC This is the way it should look uncommented. Now save it. Lol Open up a command shell in the sip_router directory. Type the following gmake all mode=debug exclude_modules="" modules you will get some errors that are caused by the imcompleteness but not to worry you wont be using them .. It will should come up with ser-0.8.14_linux_i386.tar.gz you will need to do a find for this file because I have no idea where it places it so I use Konqueror to locate it. Then do a copy command to the Home directory so you have it for later also. You will need to do some reading in the manual of SER so you will be familiar with the system. But here is a working ser.cfg using MySQL for accounting You will need to create the data base using the shell script in the SBIN directory. If you use serctl start and the pid file is not found then that means that there is an error in the ser.cfg Yes I know that the code looks ugly however so does your dog ... just kidding it is a work in progress as I work it down more I hope that this will help a few folks out. # This is a reworked script from someone else. # $Id: ser.cfg A working MySQL, Accounting PSTN Reworked for 8.14 # By Peter Bjorklund / Mr. B .. may the code gods be pleased .. lol # # You will need the adjust the ports to suit you needs # # xxx.xxx.xxx.xxx this is your IP address - ex. 123.456.789.123 # yyy.yyy.yyy.yyy this is your PSTN GateWay address ex. 123.456.789.123 # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) memlog=3 listen=xxx.xxx.xxx.xxx # this is your IP address sip_warning=yes /* Uncomment these lines to enter debugging mode debug=8 fork=no log_stderror=yes */ check_via=yes # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- 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/acc.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/exec.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/print.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri.so" # ----------------- setting module-specific parameters --------------- # -- tm params -- modparam("tm", "fr_timer", 10 ) modparam("tm", "fr_inv_timer", 24 ) modparam("tm", "wt_timer", 10 ) # modparam("tm", "uac_from", "sip:daemon@iptel.org" ) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # * 0 -- dont use mysql, 1 -- write_through, 2--write_back */ modparam("usrloc", "db_mode", 2) modparam("usrloc", "timer_interval", 10) modparam("usrloc", "db_url","mysql://ser:heslo@localhost/ser") modparam("group", "db_url","mysql://ser:heslo@localhost/ser") # -- auth params -- modparam("auth_db", "db_url","mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") # -- acc params -- # report ACKs too for sake of completeness -- as we account PSTN # destinations which are RR, ACKs should show up modparam("acc", "db_url","mysql://ser:heslo@localhost/ser") # modparam("acc", "log_missed_flag", 3) modparam("acc", "log_level", 1) modparam("acc", "db_flag", 1) modparam("acc", "report_ack", 0) modparam("acc", "log_fmt", "miocfsu") ############################################ # ------------------------- request routing logic ------------------- # main routing logic route{ /* ********* ROUTINE CHECKS ********************************** */ # filter too old messages if (!mf_process_maxfwd_header("10")) { log("LOG: Too many hops\n"); sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; ############# deal with local area code ################# you can replace the xxx with your area code if (uri=~"sip:xxx[2-9][0-9][0-9][0-9][0-9][0-9][0-9]@.*") { strip(3); # search_append("From: .*<sip:", "xxx"); # if (search("^To: .*<sip:xxx")) { # replace("To: .*<sip:xxx", "To: <sip:"); # }; }; ################# billing ########################### if (method=="INVITE" || method=="BYE") { setflag(1); setflag(2); # yes I use this flag for debuging }; /* ********* RR ********************************** */ /* Do strict routing if route headers present */ #if (loose_route()) { t_relay(); break; }; # do not use this format # instead just use this loose_route(); /* ********* DIVERSION ********************************** */ /* apply all diversions before we proceed with processing of requests for us */ ########## You can adjust the Diviersions to your needs ################### /* IM gateway diversions */ #if (uri=~"sip:.*@icq\yourdomainname\.com" # | uri=~"sip:.*@msn\.yourdomainname\.com" # | uri=~"sip:.*@aim\.yourdomainname\.com" # | uri=~"sip:.*@yahoo\.yourdomainname\.com" ) { # append_hf("P-hint: IMGW\r\n"); # if (!t_relay("xxx.xxx.xxx.xxx", "5070")) { # sl_reply_error(); # }; # break; # }; /* divert voicemail requests */ #if (uri=~"mail\.yourdomainname\.com" | uri=~":5066"| uri=~":6060") { # sethost("yourdomainname.com"); # append_hf("P-hint: VOICEMAIL\r\n"); # if ( !t_relay("mail.iptel.org", "6060")) { # sl_reply_error(); # }; # break; # }; /* ********* RR ********************************** */ # look at whether we need record-routing; # - we need it for calls from gateways (otherwise, subsequent # requests from the other # party will attempt to contact gateway # directly through blocked ports) # - we need it for Windows Messanger's IM sessions to cross # some firewalls -- we force all MESSAGEs to go via our server # to avoid blocking port numbers (some firewalls can do # standard SIP but are puzzled by Microsoft's use of obsoleted # IM session model) # - some other places may decide to set the record-routing # flag (2 chosen) too; particularly, INVITEs to our gw if ( (src_ip==yyy.yyy.yyy.yyy & method=="INVITE") || method=="MESSAGE" || method=="INFO") { setflag(2); }; /* ********* check for requests targeted out of our domain... ******* */ # sign of our domain: there is @ (username), : # (nothing) or . (host) in front of our domain name ; # if none of these cases matches, proceed with proessing of # outbound requests in route[2] if (!(uri=~"[@:\.]yourdomainname\.com([;:].*)*" # ... some phones put IP address in URI instead ... | uri=~"[@:\.]xxx\.xxx\.xxx\.xxx([;:].*)*" # ... and we serve our gateway too (we RR requests to it, so that # its address may show up in subsequent requests after # rewriteFromRoute | uri=~"@xxx\.xxx\.xxx\.xxx([;:].*)*" )) { route(2); break; }; /* ************ requests for our domain ********** */ /* now, the request is for sure for our domain */ # registers always MUST be authenticated to # avoid stealing incoming calls if (method=="REGISTER") { # Make sure that user's dont register infinite loops # (note: does not match with folded lines) if (search("^(Contact|m): .*(a)(xxx\.xxx\.xxx\.xxx|yourdomainname\.com)")) { log(1, "LOG: alert: someone trying to set aor==contact\n"); sl_send_reply("476", "No Server Address in Contacts Allowed" ); break; }; # prohibit attempts to grab someone else's To address # using valid grp; the only exception is the user # 'replciator' permitted to generate 3-rd party registrations if (!www_authorize("yourdomainname.com" /* realm */, "subscriber" /* table name */ )) { # challenge if none or invalid grp www_challenge( "yourdomainname.com" /* realm */, "0" /* no qop -- some phones can't deal with it */); break; }; if (!is_user("replicator") & !check_to()) { log("LOG: To Cheating attempt\n"); sl_send_reply("403", "That is ugly -- use To=id next time"); break; }; # it is an authenticated request, update Contact database now if (!save("location")) { sl_reply_error(); }; /* XXX not tested yet t_replicate("bat.iptel.org", "5060"); */ break; }; ###############deny calls without autorization################# if (!(src_ip==xxx.xxx.xxx.xxx | src_ip==yyy.yyy.yyy.yyy| method==ACK | method=="CANCEL" )) { if (!proxy_authorize( "yourdomainname.com" /* realm */, "subscriber" /* table name */)) { proxy_challenge( "yourdomainname.com" /* realm */, "0" /* no qop */ ); break; # let's check from=id ... avoids accounting confusion } else if (method=="INVITE" & !check_from()) { log("LOG: From Cheating attempt\n"); sl_send_reply("403", "That is ugly -- use From=id next time (gw)"); break; }; }; /* some UACs might be fooled by Contacts our UACs generate to make MSN happy (web-im, e.g.) -- tell its urneachable */ if (uri=~"sip:daemon@" ) { sl_send_reply("410", "daemon is gone"); break; }; # various aliases (might use a database in future) lookup("aliases"); # check again, if it is still for our domain after aliases # we have to include '.iptel.org' not to proceed to outbound # authentication for calls to mail.iptel.org and other hosts # served by us if ( !(uri=~"[@:\.]yourdomainname\.com([;:].*)*" | uri=~"[@:\.]xxx\.xxx\.xxx\.xxx([;:].*)*" )) { route(5); break; }; # now check if it's about PSTN destinations through our gateway; # note that 1.... and 0.... is exempted for numerical non-gw destinations if (uri=~"sip:\+?[0-1][1-9][0-9]*@.*") { route(3); break; }; # if relay if (uri=~"sip:\+?[2-9][0-9][0-9][2-9][0-9][0-9][0-9][0-9][0-9][0-9]@.*") { route(6); break; }; # does the user wish redirection on no availability? (i.e., is he # in the voicemail group?) -- determine it now and store it in # flag 4, before we rewrite the flag using UsrLoc if (is_user_in("Request-Uri", "voicemail")) { setflag(4); } ; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { # handle user which was not found ... route(4); break; }; # check whether some inventive user has uploaded gateway # contacts to UsrLoc to bypass our authorization logic if (uri=~"@yyy\.yyy\.yyy\.yyy([;:].*)*" ) { log(1, "LOG: Weird! Gateway address in UsrLoc!\n"); route(3); break; }; # if user is on-line and is in voicemail group, enable redirection # if (method=="INVITE" && isflagset(4)) { # t_on_failure("1"); # }; /* ... and also report on missed calls ... note that reporting on missed calls is mutually exclusive with silent C timer */ # setflag(3); # add RR to messages which were previously labeled for that if (isflagset(2)) { loose_route(); }; # we now know we may, we know where, let it go out now! append_hf("P-hint: USRLOC\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } #------------------- OUTBOUND ---------------------------------------- # routing logic for outbound requests targeted out of our domain # (beware, messages to our users can end up here too: for example, # an INVITE may be UsrLoc-ed, then the other party uses outbound # proxy with r-uri=the usr_loced addredd (typically IP)) route[2] { # outbound requests are allowed only for our users -- we don't # support relaying and don't like strangers bothering us # with resolving DNS; except our gateway if (!(src_ip==yyy.yyy.yyy.yyy) & !(proxy_authorize( "yourdomainname.com" /* realm */, "subscriber" /* table name */ ))) { # ACK/CANCEL have inherently no security -- just log if # included grp are wrong and proceed if (method=="BYE" ) { log("LOG: failed outbound authentication for BYE granted\n"); } else if (method=="ACK" ) { log("LOG: failed outbound authentication for ACK granted\n"); } else if (method=="CANCEL") { log("LOG: failed outbound authentication for CANCEL granted\n"); } else { proxy_challenge("yourdomainname.com" /* realm */, "0" /* no-qop */); break; }; }; # to maintain credibility of our proxy, we check From in INVITEs to be # equal to credential id so that user john.doe does not put bill.gates # in his From; we don't do that for other requests: within a dialogue, # subsequent transactions coming from the other side will have in From # To of the original transaction to match the dialog; it may be # however different from user's default From and digest; example: # user_a sends INVITE to secretary; secretary is translated to user_b; # user_b accepts and later BYEs; to match the dialog, it puts # secretary in From and sends user_b's grp -- check_from # would fail if (!src_ip==yyy.yyy.yyy.yyy& method=="INVITE" & !check_from()) { log("LOG: From Cheating attempt\n"); sl_send_reply("403", "That is ugly -- use From=id next time (OB)"); break; }; if (isflagset(2)) { record_route(); ; }; append_hf("P-hint: OUTBOUND\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } #------- ALIASED OUTBOUND -------------------------------------------- # routing logic for inbound requests aliased outbound; unlike # with real outbound requests we do not force authentication # as these calls are server by our server and we do not want # to disqualify unathenticated request originatiors from other # domains route[5] { append_hf("P-hint: ALIASED-OUTBOUND\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } #----------------- PSTN ---------------------------------------------- # logic for calls to the PSTN route[3] { # if it is a MESSAGE pass it "as is" over to our SMS gateway # (which unfortunately lives at a different host due to # lack of serial interfaces) if (method=="MESSAGE") { # set accounting setflag(1); rewritehostport("yyy.yyy.yyy.yyy:5070"); #change to our uri port for this append_hf("P-hint: SMS\r\n"); if (!t_relay()) { sl_reply_error(); }; break; }; # continue with requests to PSTN gateway ... # the international + prefix if (uri=~"sip:\+" ) { #strip(1); prefix("000"); }; # free call destinations ... no authentication needed if ( is_user_in("credentials", "free-pstn") /* free destinations */ | uri=~"sip:0@.*") { /* local PBX */ # just log it, no authentication setflag(1); setflag(2); } else { # all other PSTN destinations only for authenticated users # (Cisco GW, which has no digest support, is authenticated # by its IP address -- that's for sure not very strong; # wth confirmed that we filter packets coming from outside # and bearing SRC IP address of our network) # we are forgiving about ACK/CANCEL as digest provides no # real security for them if (!(src_ip==yyy.yyy.yyy.yyy| method==ACK | method=="CANCEL" )) { if (!proxy_authorize("yourdomainname.com" /* realm */, "subscriber" /* table name */)) { proxy_challenge("yourdomainname.com" /* realm */, "0" /* no qop */ ); break; # let's check from=id ... avoids accounting confusion } else if (method=="INVITE" & !check_from()) { log("LOG: From Cheating attempt\n"); sl_send_reply("403", "That is ugly -- use From=id next time (gw)"); break; }; }; # authorize only for INVITEs -- RR/Contact may result in weird # things showing up in d-uri that would break our logic; our # major concern is INVITE which causes PSTN costs anyway if (method=="INVITE") { # does the authenticated user have a permission for local # calls? (i.e., is he in the "local" group?) Replace xxx with your area code if (uri=~"sip:xxx[2-9][0-9]+@.*") { if (!is_user_in("credentials","local")) { sl_send_reply("403", "Local Toodle Noodle..."); break; }; # the same for long-distance } else if (uri=~"sip:1[2-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9][0-9]@.*") { #Prevent 1 900 calls etc #if (uri=~"sip:001[089]" | uri=~"sip:00900.*" ) { # sl_send_reply("403", "Added Value Destinations not permitted..."); # break; #}; if (!is_user_in("credentials","ld")) { sl_send_reply("403", "LD Toodle Noodle..."); break; }; # the same for international calls } else if (uri=~"sip:011[1-9][0-9]+@.*") { if (!is_user_in("credentials","int")) { sl_send_reply("403", "International Toodle Noodle..."); break; }; # everything else (e.g., interplanetary calls) is denied } else { sl_send_reply("403", "interplanetary Toodle Noodle..."); break; }; }; # INVITE to authorized PSTN # we passed all authorization checks for PSTN -- move on! # tag this transaction for accounting setflag(1); setflag(2); }; # authorized PSTN # requests to gateway must be record-routed because the GW accepts # only reqeusts coming from our proxy # if (isflagset(2) || method=="INVITE") if ((method=="INVITE") || (method=="ACK") || (method=="BYE")) setflag(1); # setflag(2); record_route(); # if you have passed through all the checks, let your call go to GW! # if (search('^(From|m): ".*" <sip:[2-9][0-9][0-9][0-9][0-9][0-9][0-9]@.*')) { # search_append('From: ".*" <sip:', "204"); # }; #voiplist gateway toll free only if ( uri=~"sip:1800[2-9][0-9]+@.*" | uri=~"sip:1888[2-9][0-9]+@.*" | uri=~"sip:1877[2-9][0-9]+@.*" | uri=~"sip:1866[2-9][0-9]+@.*" ) { # prefix("123456"); # replace number if your gateway uses a prefix number rewritehostport("yyy.yyy.yyy.yyy:5060"); } #voiplist LD else if ( uri=~"sip:1[1-9][0-9]*@.*" ) { # prefix("123456"); rewritehostport("yyy.yyy.yyy.yyy:5060"); } #voiplist INTERNATIONAL else if ( uri=~"sip:011[0-9][0-9]*@.*" ) { # prefix("123456"); rewritehostport("yyy.yyy.yyy.yyy:5060"); } else { sl_send_reply("403", "interplanetary Toodle Noodle..."); break; }; append_hf("P-hint: GATEWAY\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } /* *********** handling of unavailable user ******************* */ /* handling of users who are off-line */ route[4] { # user not found -- act as stateful UAS to avoid reporting # on each INVITE retranmission if (method=="INVITE" || method=="ACK" || method=="BYE" || method=="CANCEL" ) { /* requests to voicemail users will be fwded to voicemail */ if (isflagset(4)) { rewritehostport("mail.iptel.org:6060"); append_hf("P-hint: OFFLINE-VOICEMAIL\r\n"); if (!t_relay()) { sl_reply_error(); }; } else { /* non-voicemail users get 404 */ if (t_newtran()) { if (method=="ACK") { log("oops -- ACK to a non-existent transaction"); drop; }; # we reply statefuly to avoid accounting of all # retransmissions if (!t_reply("404", "Not Found")) { sl_reply_error(); }; } else { sl_reply_error(); }; }; /* we account missed calls for all off-line users */ # if (method=="INVITE") acc_request("404 Not Found"); # break; }; # non-VoIP messages: just return 404 statelessly sl_send_reply("404", "Not Found"); } ####################################### #----------------- VOIPLD --------------- # logic for calls to the VOIPLD route[6] { # if it is a MESSAGE pass it "as is" over to our SMS gateway if (method=="MESSAGE") { # set accounting setflag(1); rewritehostport("yyy.yyy.yyy.yyy:5060"); append_hf("P-hint: SMS\r\n"); if (!t_relay()) { sl_reply_error(); }; break; }; # continue with requests to PSTN gateway ... # the international + prefix if (uri=~"sip:\+" ) { #strip(1); prefix("000"); }; # free call destinations ... no authentication needed if ( is_user_in("credentials", "free-pstn") /* free destinations */ | uri=~"sip:0@.*") { /* local PBX */ # just log it, no authentication setflag(1); } else { if (!(src_ip==yyy.yyy.yyy.yyy| method==ACK | method=="CANCEL" )) { if (!proxy_authorize( "yourdomainname.com" /* realm */, "subscriber" /* table name */)) { proxy_challenge( "yourdomainname.com" /* realm */, "0" /* no qop */ ); break; } else if (method=="INVITE" & !check_from()) { log("LOG: From Cheating attempt\n"); sl_send_reply("403", "That is ugly -- use From=id next time (gw)"); break; }; }; setflag(1); }; # authorized PSTN if (isflagset(2) || method=="INVITE") record_route(); rewritehost("yyy.yyy.yyy.yyy:5060"); append_hf("P-hint: GATEWAY\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } _________________________________________________________________ Powerful Parental Controls Let your child discover the best the Internet has to offer. http://join.msn.com/?pgmarket=en-ca&page=byoa/prem&xAPID=1994&DI=1034&SU=ht… Start enjoying all the benefits of MSN® Premium right now and get the first two months FREE*.

1 0

RE: [Serusers] DNS Support for SIP SRV records - RedHat Linux9.0
by innovation.interops＠wipro.com 30 Jan '05

30 Jan '05

Hello all, I was able to make SRV query work with SER and bind 9x. I was trying to use this pattern in bind zone file _sip._udp.testdomain.com which could be the reason why it rejected. I modified to simple _sip.udp SRV 0 0 5060 server.testdomain.com and it started working.I did not know the reason. Also why donot I see any NAPTR query from SER?? Regds karthikeyan.k ________________________________ From: Steve Blair [mailto:blairs@isc.upenn.edu] Sent: Thu 1/27/2005 5:10 PM To: Innovation Interops Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] DNS Support for SIP SRV records - RedHat Linux9.0 Can query manually query the DNS server for the SRV record? If so what happens? innovation.interops(a)wipro.com wrote: > Hello all, > > Have any one setup SIP srv records in Red Hat Linux 9.0 to work with SER. > > I tried adding a SIP UDP based srv record in the zone file but when > SER queries for the SRV record the DNS is replying not found even > though it is available in the zone file. > > I am using Bind 9x. > > > Regds > karthikeyan.k > > > Confidentiality Notice > > 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 confidential > or privileged information. If > you are not the intended recipient, please notify the sender at Wipro > or Mailadmin(a)wipro.com immediately > and destroy all copies of this message and any attachments. > >------------------------------------------------------------------------ > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers > > -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:blairs@upenn.edu Confidentiality Notice 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 confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments.

6 5

[Serusers] serusers as newsgroup ?
by Hans-Henrik Andresen 30 Jan '05

30 Jan '05

Hi Can I get the serusers mail as newsgroup anywhere ? /HH _________________________________________________________________ Is your PC infected? Get a FREE online computer virus scan from McAfee® Security. http://clinic.mcafee.com/clinic/ibuy/campaign.asp?cid=3963

1 0

[Serusers] accounting problem if calls not completed
by Mohamed Omar 30 Jan '05

30 Jan '05

I'm running SER 8.14 with radius and accounting. Accounting is working fine when the calls are completed but If I make a call and I get a status 404. no accounting information is sent to the radius server. I need this information sent to the radius server as my billing server opens a session for every invite for a PSTN call and will only close the session when accounting information is recieved from the radius server. How can I configure SER to flag status 404 for accounting and close the transaction? Thanks in Advance Mohamed. --------------------------------- Post your free ad now! Yahoo! Canada Personals

1 0

[Serusers] You are made active
by Jiri 30 Jan '05

30 Jan '05

1 0

← Newer
1
2
3
4
5
6
7
...
40
Older →

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users January 2005