- sr-users - kamailio.org

[Serusers] ser.cfg.errors
by Ashling O'Driscoll 31 Jan '05

31 Jan '05

I was hoping someone could help identify why I have errors in my config file. When I try to start SER,I am told that it has failed and there are three erros in my config file. When I comment out the following line I can get rid of one error: #loadmodule "/usr/local/lib/ser/modules/mysql.so" I dont even understand why that is an error. Please help me to understand why my config file is incorrect. Thank you. # # $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) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/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" # 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 > max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("xxx.xxx.xxx.xxx", "subscriber")) { www_challenge("xxx.xxx.xxx.xxx", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } -------------------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.

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[Serusers] (no subject)
by Felipe Martins 31 Jan '05

31 Jan '05

Hi Everybody, I'm trying to make SER work with ASTERISK, ser as a proxy server and SIP terminator, and asterisk for other services like voicemail and International calls like an UAC (thing that I know SER is not able to do). I've configured SER with the following forward route, to forward all "1" beggining call phones to forward to my asterisk for international calls: if (uri=~"^sip:8[0-9]{7}.*") { forward( 10.0.18.3, 5061 ); break; } When I restart SER, it gives me this errors: Jul 22 12:49:54 sipproxy /usr/local/sbin/ser[479]: ERROR: send_rtpp_command: can't connect to RTP proxy Jul 22 12:49:54 sipproxy /usr/local/sbin/ser[479]: WARNING: rtpp_test: can't get version of the RTP proxy Jul 22 12:49:54 sipproxy /usr/local/sbin/ser[479]: WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily I've also tried with the following rewritehost but it gives me the same error: if (uri=~"^sip:8[0-9]{7}.*") { rewritehost("10.0.18.3, 5061"); break; } I know that i'm missing something with RTP that i'm still working on. I've tried to look for it at google, but find nothing interesting. Do anyone know what is going on ? My ser.cfg is attached. -- Felipe Martins Linux System Administrator Tep Solution Provider Mundivox Communications Rua Lauro Muller, 116/Sala 505 RJ - Brasil - 22290-906 Tel.: 55 21 3820-8839 Fax.: 55 21 3820-8844

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[Serusers] Getting contact URI
by Tom Gaudasinski 31 Jan '05

31 Jan '05

Greetings, I have two questions. What function() would i use to get a bool back indicating whether a certain user is in the usrloc given i only know the 'username' field of the usrloc table? Secondly how can I get the contact URI/IP address of a user given again I only know the 'username' field of the usrloc table?; this will be used in a forward() statement. Thank you.

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[Serusers] Internal numbers prefix
by Pavel Siderov - Hostmates 31 Jan '05

31 Jan '05

Hi there, I'm trying to forward some destinations, but there are accounts/users starting the same way connected to the server. Example account: 123456 When ser see that it forwards the call to other ser located at 12.34.56.79. I want my users to use prefix for local calls. Example: calling 699123456 goes to local server user 123456 not to 12.34.56.79. Please check my code and tell me what's wrong: ... if (uri=~"^sip:1234") { log(0, "\n\n************ Office 1 ***********\n"); strip(0); rewritehostport("12.34.56.79:5060"); append_branch("12.34.56.79"); t_relay_to_udp("12.34.56.79", "5060"); append_hf("P-hint: VoipSwitch GATEWAY\r\n"); log(0, "********************************************\n\n\n"); break; }; if (uri=~"^sip:051") { log(0, "\n\n************ Office 2 ***********\n"); strip(0); rewritehostport("12.34.56.78:5060"); append_branch("12.34.56.78"); t_relay_to_udp("12.34.56.78", "5060"); append_hf("P-hint: VoipSwitch GATEWAY\r\n"); log(0, "********************************************\n\n\n"); break; }; if (uri=~"^sip:699") { strip(3); route(1); break; }; } else { sl_send_reply("399", "Your account balance is low!"); }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { if (isflagset(6)) { force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; } onreply_route[1] { if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); } else if (nat_uac_test("1")) { fix_nated_contact(); }; } Thanks in advance :)! Pavel Siderov

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[Serusers] Linksys PAP2 problem
by ron 31 Jan '05

31 Jan '05

Hi All, Anyone here uses the Linksys PAP2-NA(open version)? I'm having problem right now because I have the 2-port Linksys PAP2-NA. The problem is that when one port(any of the ports) is being use to call pstn the other port cannot make any more calls, meaning they can make class one at a time only, this occurs only when they call pstn, but when they are calling local extensions on our sip server, no problem, both ports works simultaneously. I did a packet sniff, I'm seeing that the gateway is sending service unavailble to my SIP server. Is this a problem in the Linksys, gateway or the SIP server itself? Thank you in advanced regards, Ron -- -- iTouch Communications, Inc.

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[Serusers] multi listen port
by janakl janakl 31 Jan '05

31 Jan '05

Hi, Does ser can listen several different port? thanks. _________________________________________________________________ 与联机的朋友进行交流，请使用 MSN Messenger: http://messenger.msn.com/cn

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[Serusers] Accounting Question on 404 status code
by Mohamed Omar 31 Jan '05

31 Jan '05

How can make SER flags status 404 for accounting and send a stop accouting record to the radius server? Am currently doing accounting for calls going to PSTN network and it's working fine if the call is completed but if I get an error like status 404 then no stop accounting record is sent to the radius server. Thanks. Mohamed --------------------------------- Post your free ad now! Yahoo! Canada Personals

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[Serusers] Seeking advice about : PDT module.
by Ricardo Martinez 30 Jan '05

30 Jan '05

Hello list. Regarging to this module i have a question. i want to implement a "domain" list with all the codes, or at least as many as i can use. The problem is that i can't have a "terminating" digit for every code. The way a user dial a number in my system is something like : 00 + Country Code + City Code + Number. I want to match the Country Code + City Code. but I can't add a "terminating" digit for every call or even more, force to the enduser to add a "terminating" digit before the Country Code + City Code when he's dialing, that only will add more confusion. My question is : Is possible to avoid or bypass the "terminating digit"? and maybe have a match with the best code?. 56+2+1234567 will match with 56+2 56+9+1235678 will match with 56+9 and so on..... Hope that someone could give me a hand here. Ricardo Martinez Ogalde.

2 2

[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

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[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

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[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*.

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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

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[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

[Serusers] Email Check
by Mr. B 30 Jan '05

30 Jan '05

Sorry, I sent a fairly large email to the group and I am not sure if it got bounced. It was a mini how to . this email is just to see if my @#$%@#$%$ server is working. LOL

1 0

[Serusers] Accounting, Logging, PSTN, MySQL a Mini HowTo
by Mr. B 30 Jan '05

30 Jan '05

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 . don't 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 iso's (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 won't 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 */ /* 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; }; }

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[Serusers] Problems with Mysql - Resend
by ranganathu＠canind-inc.com 29 Jan '05

29 Jan '05

Hi, I am a newbie to SER. Just today I successfully set up and terminated a call between users using a SER server running on a P4 MandrakeLinux 10.1 machine. I used the ser.cfg script that came along with the installation. This, I understand does not allow persistent users. Since I had built my installation with Mysql modules, I set up mysql to run along with ser by executing the ser_mysql.sh. It went through fine and I saw that the tables were all created too. I modified the ser.cfg file so as to include user authentication through mysql. My ser.cfg file is found attached herewith. As and when I restart ser, this is the error message that I get: ------------- 0(7118) loading module /usr/local/lib/ser/modules/mysql.so 0(7118) ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: libmysqlclient.so.10: cannot open shared object file: No such file or directory 0(7118) parse error (29,13-49): failed to load module 0(7118) loading module /usr/local/lib/ser/modules/sl.so 0(7118) loading module /usr/local/lib/ser/modules/tm.so 0(7118) loading module /usr/local/lib/ser/modules/rr.so 0(7118) loading module /usr/local/lib/ser/modules/maxfwd.so 0(7118) loading module /usr/local/lib/ser/modules/usrloc.so 0(7118) loading module /usr/local/lib/ser/modules/registrar.so 0(7118) loading module /usr/local/lib/ser/modules/auth.so 0(7118) loading module /usr/local/lib/ser/modules/auth_db.so 0(7118) loading module /usr/local/lib/ser/modules/acc.so 0(7118) loading module /usr/local/lib/ser/modules/exec.so 0(7118) loading module /usr/local/lib/ser/modules/group.so 0(7118) loading module /usr/local/lib/ser/modules/msilo.so 0(7118) loading module /usr/local/lib/ser/modules/print.so 0(7118) loading module /usr/local/lib/ser/modules/textops.so 0(7118) loading module /usr/local/lib/ser/modules/uri.so 0(7118) loading module /usr/local/lib/ser/modules/vm.so 0(7118) set_mod_param_regex: usrloc matches module usrloc 0(7118) set_mod_param_regex: found <db_mode> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(7118) set_mod_param_regex: auth_db matches module auth_db 0(7118) set_mod_param_regex: found <calculate_ha1> in module auth_db [/usr/local/lib/ser/modules/auth_db.so] 0(7118) set_mod_param_regex: auth_db matches module auth_db 0(7118) set_mod_param_regex: found <password_column> in module auth_db [/usr/local/lib/ser/modules/auth_db.so] 0(7118) set_mod_param_regex: rr matches module rr 0(7118) set_mod_param_regex: found <enable_full_lr> in module rr [/usr/local/lib/ser/modules/rr.so] 0(7118) find_export: found <mf_process_maxfwd_header> in module maxfwd_module [/usr/local/lib/ser/modules/maxfwd.so] 0(7118) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(7118) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(7118) find_export: found <record_route> in module rr [/usr/local/lib/ser/modules/rr.so] 0(7118) find_export: found <loose_route> in module rr [/usr/local/lib/ser/modules/rr.so] 0(7118) find_export: found <t_relay> in module tm [/usr/local/lib/ser/modules/tm.so] 0(7118) find_export: found <www_authorize> in module auth_db [/usr/local/lib/ser/modules/auth_db.so] 0(7118) find_export: found <www_challenge> in module auth [/usr/local/lib/ser/modules/auth.so] 0(7118) find_export: found <save> in module registrar [/usr/local/lib/ser/modules/registrar.so] 0(7118) find_export: found <lookup> in module registrar [/usr/local/lib/ser/modules/registrar.so] 0(7118) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(7118) find_export: found <t_relay> in module tm [/usr/local/lib/ser/modules/tm.so] 0(7118) find_export: found <sl_reply_error> in module sl_module [/usr/local/lib/ser/modules/sl.so] ERROR: bad config file (1 errors) 0(7118) MSILO: destroy module ... 0(7118) DEBUG: tm_shutdown : start 0(7118) DEBUG: tm_shutdown : empting hash table 0(7118) DEBUG: tm_shutdown: releasing timers 0(7118) DEBUG: tm_shutdown : removing semaphores 0(7118) DEBUG: tm_shutdown : done 0(7118) shm_mem_destroy 0(7118) destroying the shared memory lock ------------------ Is this because I am using a different version of mysql and it cannot find libmysqlclient.so.10? When I did the compilation I did it so with libmysqlclient.so.12 and assumed that everything would be taken care of. Any suggestions would be wonderful. Thanks, Ranganath -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

1 0

[Serusers] SEMS MP3 help
by Steve Blair 29 Jan '05

29 Jan '05

I am trying to re-build a copy of sems with te lame mp3 encoder. The build seems to complete without errors but when I try to start sems I get "sems[2560]: Error: audio plug-in doesn't contain any exports ! " error and sems doesn't start. Any idea why? Any idea how to correct this? Thanks,Steve

1 0

[Serusers] (no subject)
by ranganathu＠canind-inc.com 29 Jan '05

29 Jan '05

Hi, I am a newbie to SER. Just today I successfully set up and terminated a call between users using a SER server running on a P4 MandrakeLinux 10.1 machine. I used the ser.cfg script that came along with the installation. This, I understand does not allow persistent users. Since I had built my installation with Mysql modules, I set up mysql to run along with ser by executing the ser_mysql.sh. It went through fine and I saw that the tables were all created too. I modified the ser.cfg file so as to include user authentication through mysql. My ser.cfg file is found attached herewith. As and when I restart ser, this is the error message that I get: ------------- 0(7118) loading module /usr/local/lib/ser/modules/mysql.so 0(7118) ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: libmysqlclient.so.10: cannot open shared object file: No such file or directory 0(7118) parse error (29,13-49): failed to load module 0(7118) loading module /usr/local/lib/ser/modules/sl.so 0(7118) loading module /usr/local/lib/ser/modules/tm.so 0(7118) loading module /usr/local/lib/ser/modules/rr.so 0(7118) loading module /usr/local/lib/ser/modules/maxfwd.so 0(7118) loading module /usr/local/lib/ser/modules/usrloc.so 0(7118) loading module /usr/local/lib/ser/modules/registrar.so 0(7118) loading module /usr/local/lib/ser/modules/auth.so 0(7118) loading module /usr/local/lib/ser/modules/auth_db.so 0(7118) loading module /usr/local/lib/ser/modules/acc.so 0(7118) loading module /usr/local/lib/ser/modules/exec.so 0(7118) loading module /usr/local/lib/ser/modules/group.so 0(7118) loading module /usr/local/lib/ser/modules/msilo.so 0(7118) loading module /usr/local/lib/ser/modules/print.so 0(7118) loading module /usr/local/lib/ser/modules/textops.so 0(7118) loading module /usr/local/lib/ser/modules/uri.so 0(7118) loading module /usr/local/lib/ser/modules/vm.so 0(7118) set_mod_param_regex: usrloc matches module usrloc 0(7118) set_mod_param_regex: found <db_mode> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(7118) set_mod_param_regex: auth_db matches module auth_db 0(7118) set_mod_param_regex: found <calculate_ha1> in module auth_db [/usr/local/lib/ser/modules/auth_db.so] 0(7118) set_mod_param_regex: auth_db matches module auth_db 0(7118) set_mod_param_regex: found <password_column> in module auth_db [/usr/local/lib/ser/modules/auth_db.so] 0(7118) set_mod_param_regex: rr matches module rr 0(7118) set_mod_param_regex: found <enable_full_lr> in module rr [/usr/local/lib/ser/modules/rr.so] 0(7118) find_export: found <mf_process_maxfwd_header> in module maxfwd_module [/usr/local/lib/ser/modules/maxfwd.so] 0(7118) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(7118) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(7118) find_export: found <record_route> in module rr [/usr/local/lib/ser/modules/rr.so] 0(7118) find_export: found <loose_route> in module rr [/usr/local/lib/ser/modules/rr.so] 0(7118) find_export: found <t_relay> in module tm [/usr/local/lib/ser/modules/tm.so] 0(7118) find_export: found <www_authorize> in module auth_db [/usr/local/lib/ser/modules/auth_db.so] 0(7118) find_export: found <www_challenge> in module auth [/usr/local/lib/ser/modules/auth.so] 0(7118) find_export: found <save> in module registrar [/usr/local/lib/ser/modules/registrar.so] 0(7118) find_export: found <lookup> in module registrar [/usr/local/lib/ser/modules/registrar.so] 0(7118) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(7118) find_export: found <t_relay> in module tm [/usr/local/lib/ser/modules/tm.so] 0(7118) find_export: found <sl_reply_error> in module sl_module [/usr/local/lib/ser/modules/sl.so] ERROR: bad config file (1 errors) 0(7118) MSILO: destroy module ... 0(7118) DEBUG: tm_shutdown : start 0(7118) DEBUG: tm_shutdown : empting hash table 0(7118) DEBUG: tm_shutdown: releasing timers 0(7118) DEBUG: tm_shutdown : removing semaphores 0(7118) DEBUG: tm_shutdown : done 0(7118) shm_mem_destroy 0(7118) destroying the shared memory lock ------------------ Is this because I am using a different version of mysql and it cannot find libmysqlclient.so.10? When I did the compilation I did it so with libmysqlclient.so.12 and assumed that everything would be taken care of. Any suggestions would be wonderful. Thanks, Ranganath -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

1 0

Re: [Serusers] Looking for SER + Asterisk-as-voicemail HOWTO
by Ashling O'Driscoll 29 Jan '05

29 Jan '05

Thanks for the reply Girish. I would just like to verify the following: I currently have the following very simple config: if (method == "INVITE") { //other code t_on_failure("1"); }; failure_route[1]{ revert_uri(); rewritehostport("xxx.xxx.xxx.xxx:5062"); append_branch(); t_relay_on_udp("xxx.xxx.xxx.xxx","5062"); break; } However when I restart SER it tells me I have an error in my ser.cfg and from the log file it says I am missing a loadmodule: The following are the load modules I have inclued. Am I missing one? sl.so tm.so rr.so maxfwd.so usrloc.so registrar.so nathelper.so textops.so auth.so authdb.so Thanks, Aisling. ---- Original Message ---- From: gr_sh2003(a)yahoo.com To: ashling.odriscoll(a)cit.ie, serusers(a)iptel.org Subject: Re: [Serusers] Looking for SER + Asterisk-as-voicemail HOWTO Date: Sat, 29 Jan 2005 00:39:19 -0800 (PST) >Ashling, > >See comments inline. > >--- Ashling O'Driscoll <ashling.odriscoll(a)cit.ie> wrote: >> >> I implemented the changes in ser.cfg that Giri suggested below(will >> this cover no answer and busy?). Asterisk and ser are on the same >> machine so I now forward to 5062: see code below. Hoever how do I >now >> modify Asterisk to handle this?...Like I said Ive read lots on this >> but it gets complicated with perl scripts etc. Apologies if I am a >> bit slow picking up on this! > >Go thru the README.variables file that comes with the Asterisk >distribution. It describes about >the extension logic and how to handle different variables. Also, >learn about pattern matching for >Asterisk extensions. Now add appropriate entries for voicemail in >voicemail.conf. Have something >like this in your extensions.conf when the vm calls from SER reach >Asterisk. > >exten => _X.,1,VoiceMail2(u${EXTEN}) > >> >> ser.cfg; >> if(!lookup("location"){ >> rouet(2); >> break; >> }; > > >The above code will not serve your busy/unanswered needs. Add a >failure route and handle it there. >like: > >t_on_failure("1"); >t_relay (); >break; > >failure_route[1] { > route(2); > break; >} > >> route [2] >> { >> rewritehostport("xxx.xxx.xxx.xxx:5062"); >> t_relay_to_udp("xxx.xxx.xxx.xxx:5062"); >> break; >> }; >> >> Aisling > >Also, learning about using t_check_status will also help you handle >specific error conditions. > >Best, > >===== >Girish Gopinath <gr_sh2003(a)yahoo.com> > >__________________________________________________ >Do You Yahoo!? >Tired of spam? Yahoo! Mail has the best spam protection around >http://mail.yahoo.com > >-------------------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. > -------------------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.

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[Serusers] SERWeb latest version: DB tables resp. fields not found....
by Alexander Philipp Lintenhofer 29 Jan '05

29 Jan '05

I installed ser09 with mysql.so and acc.so and tried to implement serweb (latest version fetched by CVS this night). Unfortunately, the generated SQL statements do not fit to the actual implementation of the database: log errors for example: Unknown column 't1.caller_deleted' in 'where clause' Table 'ser09.uuidaliases' doesn't exist I use Slackware 10 with kernel 2.6.10, PHP 4.3.9 (5.0.3 did not work with serweb) and MySQL 4.1.8 and I know that the DB-model of ser09 differs from the model of ser0814. Where can I therefore get the latest version of the DB-model ? kind regards and thank you for your help! Philipp

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