sr-users January 2004

sr-users@lists.kamailio.org

142 participants
338 discussions

Re: [Serusers] CPLEd and cpl-c module
by Brinker, Oliver 30 Jan '04

30 Jan '04

Hi Bogdan, better late then never ;) The upload of CPL scripts via SIP works fine with CPLEd but after changing the SER config file with the if statement you mentioned below I cannot REGISTER my client any more. It is a normal REGISTER without any payload in the SDP part. I try it with three different SIP clients and they all get a 500 error every time. Why do normal REGISTER messages run into the if statement, too? Many thanks for your help. Regards, Oliver Bogdan-Andrei IANCU wrote >Hi Oliver, > >Instead of using HTTP for upload/download (which requires more pieces >and more complex senttings) I would recommend you to use directly SIP >for transporting CPL scripts (via REGISTER). CPLEd supprts also this method. >In order to separated normal REGISTERs from CPL transport REGISTERS, > From the SER part you must do something like: > >if (method=="REGISTER") { > if ( cpl_process_register() ) { > sl_send_reply("500","Error"); > }; > # here you can do normal processing of a > # REGISTER request or whatever you want >}; > >In CPLEd, in the upload/download window you have to fill in the address >and port of SER and the user name for which you want to install/fetch >the script. > >Regards, >Bogdan Brinker, Oliver wrote: >Hi Daniel, >when I try to upload a cpl script via http the status string of CPLEd >displays the following error message: > >Error: Upload via HTTP: HTTP request failed: Invalid CPL script > >I've set the direct path to the cpl.dtd in the cpl.php and in the CPLEd >settings. Although Java SDK 1.4.1 is installed on the machine I load >crimson.jar and xalan.jar into the classpath. But what is checker.jar >and where can I get it? Is jaxp.jar included in the Java SDK or should >I download a separate version? > >Many thanks, >Oliver > > >Daniel-Constantin Mierla wrote > > > >>Hello, >>if you use the CVS head version then you should change the format of >>database url: Now it reflects the type of database used and for mysql >>should be like: >> >>mysql://foo:bar@localhost/ser >> >>Try with this one when setting the "cpl_db" parameter. >> >>Daniel >> >> > > > >>On 11/26/2003 11:30 AM, Brinker, Oliver wrote: >> >> > > > >>Hi Bogdan, >> >>The mysql module is loaded before the cpl-c one. Please have a look into >>my ser.cfg because I cannot find any misconfig. Could it be a problem >>with the SQL connection string? MySQL should be running correct. >> >>I change the module name in the modparam declaration from cpl_c to cpl-c >>because cpl_c like in your documentation doesn't work ;) Thanks >> >>Oliver >> >># >># $Id: ser.cfg,v 1.24 2003/11/11 15:32:36 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 >>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/cpl-c.so" >>loadmodule "/usr/local/lib/ser/modules/sl.so" >>loadmodule "/usr/local/lib/ser/modules/tm.so" >>loadmodule "/usr/local/lib/ser/modules/rr.so" >>loadmodule "/usr/local/lib/ser/modules/maxfwd.so" >>loadmodule "/usr/local/lib/ser/modules/usrloc.so" >>loadmodule "/usr/local/lib/ser/modules/registrar.so" >>loadmodule "/usr/local/lib/ser/modules/textops.so" >> >># 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) >> >># -- cpl-c module -- >>modparam("cpl-c","cpl_db","sql://foo:bar@localhost/ser") >>modparam("cpl-c","cpl_table","cpl") >>modparam("cpl-c","cpl_dtd_file","/usr/local/etc/ser/cpl-06.dtd") >>modparam("cpl-c","log_dir","/var/log/ser/cpl") >>modparam("cpl-c","proxy_recurse",0) >> >># ------------------------- 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 >> if (!method=="REGISTER") record_route(); >> >> # subsequent messages withing a dialog should take the >> # path determined by record-routing >> if (loose_route()) { >> # mark routing logic in request >> append_hf("P-hint: rr-enforced\r\n"); >> route(1); >> break; >> }; >> >> if (!uri==myself) { >> # mark routing logic in request >> append_hf("P-hint: outbound\r\n"); >> route(1); >> break; >> }; >> >> # if the request is for other domain use UsrLoc >> # (in case, it does not work, use the following command >> # with proper names and addresses in it) >> if (uri==myself) { >> >> if (method=="REGISTER") { >> >> # Uncomment this if you want to use digest authentication >> if (!www_authorize("t-systems.com", "subscriber")) { >> www_challenge("t-systems.com", "0"); >> break; >> }; >> >> save("location"); >> break; >> }; >> >> lookup("aliases"); >> if (!uri==myself) { >> append_hf("P-hint: outbound alias\r\n"); >> route(1); >> break; >> }; >> >> # native SIP destinations are handled using our USRLOC DB >> if (!lookup("location")) { >> sl_send_reply("404", "Not Found"); >> break; >> }; >> }; >> append_hf("P-hint: usrloc applied\r\n"); >> route(1); >>} >> >>route[1] >>{ >> # send it out now; use stateful forwarding as it works reliably >> # even for UDP2TCP >> if (!t_relay()) { >> sl_reply_error(); >> }; >>} >> >> >>Bogdan-Andrei IANCU [mailto:iancu@fokus.fraunhofer.de] wrote >> >> >> >> >> >>>Hi Oliver, >>> >>>The mysql module must be loade before the cpl-c one. Make sure about >>>this in your configuration file. >>> >>>Bogdan >>> >>> >>> >>> >>Brinker, Oliver wrote: >> >> >> >> >> >>>Hi Bogdan, >>>now I fetch the whole cvs tree and compile SER and the needed modules. >>>Everything works fine until I put the cpl-c modparam values into ser.cfg. >>>I get an error message on my xconsole that the cpl-c module cannot bind >>>to the database. The mysql module is loaded because digest authentications >>>works since a few days. Is there anything to mind regarding the order of >>>loading modules or setting params in the ser.cfg? Many thanks. >>> >>>Regards, >>>Oliver >>> >>> >>>Bogdan-Andrei IANCU [mailto:iancu@fokus.fraunhofer.de] wrote >>> >>> >>> >>> >>> >>> >>> >>>>Hi Oliver, >>>> >>>>what branch of cvs do you use? or you use the head of cvs? >>>>on CVS head everything is fixed - but you have to get the whole tree, >>>>not only the TM. >>>> >>>>Regards, >>>>Bogdan >>>> >>>> >>>> >>>> >>>> >>>> >>>Brinker, Oliver wrote: >>> >>> >>> >>> >>> >>> >>> >>>>Hi Bogdan, >>>> >>>>I try to patch the TM module and get an error during patching t_reply.c: >>>> >>>>Hunk #2 FAILED at 375 >>>>Hunk #3 FAILED at 400 >>>> >>>>I'm not sure but could the faked_env function be the problem (expect four params, >>>>but receiving only two)? Should I use a new version of the TM module from the CVS? >>>>Many thanks. >>>> >>>>Regards, >>>>Olli >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>>>Hello, >>>>> >>>>>In order to compile the cpl-c module, you need to apply a patch to the >>>>>tm_modulel you can finf this patch in modules/cpl-c/tm.patch. >>>>>Please, use the patch from the same version of ser as the tm modules. Try to do >>>>>so and let my know if it worked. >>>>> >>>>>Regards, >>>>>Bogdan >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> >>> >>> >>> >>_______________________________________________ >>Serusers mailing list >>serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> >> > > > >

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Re: [Serusers] download firmware
by Atle Samuelsen 30 Jan '04

30 Jan '04

You have all the Requierd files I hope ? OS79XX.TXT SIPDefault.cnf SIPMaxaddr.cnf and.. remember.. you have to make the upgrade step by step.. - Atle * Kuzlu <mkuzlu(a)hotmail.com> [040130 12:39]: > I move SIP firmware to TFTT directory root . > Then in 7960 write TFT server's IP address > and I see "unprovison" on screen. > > Thanks > > Kuzlu > ----- Original Message ----- > From: "Atle Samuelsen" <clona(a)camaro.no> > To: "Kuzlu" <mkuzlu(a)hotmail.com> > Cc: <serusers(a)lists.iptel.org> > Sent: Friday, January 30, 2004 12:13 PM > Subject: Re: [Serusers] download firmware > > > > > > make a call to neareast Cisco reseller, and tell them that you need all > > the firmware updates from 3.X branch to 6.X > > > > Atleast , thats what I did. > > > > - Atle > > > > > > > > * Kuzlu <mkuzlu(a)hotmail.com> [040130 11:07]: > > > How can I download a firmware file to 7960 ? > > > > > _______________________________________________ > > > Serusers mailing list > > > serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > >

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[Serusers] RE: Multiple Domain (kapil dhawan)
by kapil dhawan 30 Jan '04

30 Jan '04

Hi... i got the p[oint of using domain table i am already using it..that is to check for multiple domain verification only.....but for eg...if i add two users with username 111 with domain as a.com and b.com....it does in subscriber table with username as 111 and domains as a.com and b.com....this all is fine....when 111(a)a.com tries to call like 111#...what will happen...call only goes for last registration done...coz in location structure , it doesn't add the latest entry of 111(a)b.com.... whenever u make a call it searches for the first record in location table and call goes to it...so 111 a.com 111 b.com if the entries are like this....b.com can call a.com but not vice-versa... _________________________________________________________________ Play the prediction game on MEZ. Win Sehwags autographed T-shirts. http://go.msnserver.com/IN/41491.asp Predict and win on myenjoyzone.com.

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[Serusers] If anyone can comment of asterisk problem
by John Foster 30 Jan '04

30 Jan '04

Hi List, i tried on asterisk list as well, but any of you can give any idea abt SIP authenticatin problem with Asterisk and SIP UA (i tried SIPPS and X-Lite) I could not properly get authenticated with my SIP UA to asterisk. i m using a username for UA "12321" and following are SIP.conf file user params [12321] type=friend username=12321 host=dynamic secret=ccarta context=default mailbox=1234,2345 ; Mailbox for message waiting indicator [77777] type=friend username=77777 host=dynamic secret=atracc context=default mailbox=1234,2345 m trying with SIPPS UA, that gets status "Acquired", not "Registered", Can anyone give any idea about it? I tried same with X-Lite, didnt work. Sip debug messages are pasted below. Best Regards, JF Sip read: REGISTER sip:192.168.100.71 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992800961-40de3e5f192.168.100.66 From: <sip:12321@192.168.100.71>;tag=3b2cf0ba To: <sip:12321@192.168.100.71> Call-ID: 990209125-415b5d61@990209122-415b5d5e Contact: ccarta <sip:12321@192.168.100.66:5062>;expires=600;q=0.500 Expires: 600 CSeq: 1 REGISTER Content-Length: 0 User-Agent: Ahead SIPPS IP Phone Version 2.0.42.13 10 headers, 0 lines Using latest request as basis request Sending to 192.168.100.66 : 5062 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992800961-40de3e5f192.168.100.66 From: <sip:12321@192.168.100.71>;tag=3b2cf0ba To: <sip:12321@192.168.100.71>;tag=as648287fa Call-ID: 990209125-415b5d61@990209122-415b5d5e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:12321@192.168.100.71> Content-Length: 0 to 192.168.100.66:5062 Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992800961-40de3e5f192.168.100.66 From: <sip:12321@192.168.100.71>;tag=3b2cf0ba To: <sip:12321@192.168.100.71>;tag=as648287fa Call-ID: 990209125-415b5d61@990209122-415b5d5e CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:12321@192.168.100.71> Proxy-Authenticate: Digest realm="asterisk", nonce="7c7fba4d" Content-Length: 0 to 192.168.100.66:5062 Sip read: REGISTER sip:192.168.100.71 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992804895-411b471d192.168.100.66 From: <sip:12321@192.168.100.71>;tag=3b2d0018 To: <sip:12321@192.168.100.71> Call-ID: 990209125-415b5d61@990209122-415b5d5e Contact: ccarta <sip:12321@192.168.100.66:5062>;expires=600;q=0.500 Expires: 600 CSeq: 2 REGISTER Content-Length: 0 Proxy-Authorization: Digest username="12321",realm="asterisk",uri="sip:192.168.100.66",nonce="7c7fba4d",nc="00000001",response="b8b1d7fc53eff354dfc31dfa3f800749" User-Agent: Ahead SIPPS IP Phone Version 2.0.42.13 11 headers, 0 lines Using latest request as basis request Sending to 192.168.100.66 : 5062 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992804895-411b471d192.168.100.66 From: <sip:12321@192.168.100.71>;tag=3b2d0018 To: <sip:12321@192.168.100.71>;tag=as648287fa Call-ID: 990209125-415b5d61@990209122-415b5d5e CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:12321@192.168.100.71> Content-Length: 0 to 192.168.100.66:5062 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992804895-411b471d192.168.100.66 From: <sip:12321@192.168.100.71>;tag=3b2d0018 To: <sip:12321@192.168.100.71>;tag=as648287fa Call-ID: 990209125-415b5d61@990209122-415b5d5e CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 600 Contact: <sip:12321@192.168.100.71>;expires=600 Date: Tue, 14 Oct 2003 13:46:14 GMT Content-Length: 0 to 192.168.100.66:5062 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:12321@192.168.100.66:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.71:5060;branch=z9hG4bK3ecb7a3b From: "asterisk" <sip:asterisk@192.168.100.71>;tag=as3f6e8c0e To: <sip:12321@192.168.100.66:5062> Contact: <sip:asterisk@192.168.100.71> Call-ID: 074dfadf24b95dd75e17b56c67ffcaf1(a)192.168.100.71 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 36 Messages-Waiting: no Voicemail: 0/0 (no NAT) to 192.168.100.66:5062 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.71;branch=z9hG4bK3ecb7a3b From: "asterisk" <sip:asterisk@192.168.100.71>;tag=as3f6e8c0e To: <sip:12321@192.168.100.66:5062>;tag=3b302259 Call-ID: 074dfadf24b95dd75e17b56c67ffcaf1(a)192.168.100.71 CSeq: 102 NOTIFY User-Agent: Ahead SIPPS IP Phone Version 2.0.42.13 Content-Length: 0 8 headers, 0 lines --------------------------------- Do you Yahoo!? The New Yahoo! Shopping - with improved product search

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[Serusers] Multiple Domain
by Adrian Georgescu 30 Jan '04

30 Jan '04

# lookup local SIP domains in domain table modparam("domain", "db_mode", 1) # 0 do not use caching, 1 use caching modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") # -- usrloc params -- modparam("usrloc", "use_domain", 1) if (is_from_local()) { # this is one for our subscribers etc.... }; create a mysql table called domain, see sample scripts from ser installation you can manage the domains with serctl command > Message: 34 > Date: Fri, 30 Jan 2004 16:09:09 +0530 > From: "kapil dhawan" <oswriter(a)hotmail.com> > Subject: [Serusers] Multiple Domain > To: serusers(a)lists.iptel.org > Cc: oswriter(a)hotmail.com > Message-ID: <BAY2-F159HkfWWubeYz00013561(a)hotmail.com> > Content-Type: text/plain; format=flowed > > Hi All..... > I am not sure whether i am posting my query at right place... > > > Well i am using ser-0.8.12. i want to allow mulpile domains....but i > don't > want to do changes in .cfg file whenever a new domain is added...and i > even > don't want to write script to add location to different location > tables...depending upon the uri.....i want to use one subscriber and > one > location table.....How can this be done >

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[Serusers] download firmware
by Kuzlu 30 Jan '04

30 Jan '04

How can I download a firmware file to 7960 ?

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[Serusers] Download Firmware
by Kuzlu 30 Jan '04

30 Jan '04

How can I downlad firmware to 7960?

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[Serusers] Questions about the clients SIP cisco?
by Kuzlu 30 Jan '04

30 Jan '04

How can I download firmware to 7960?

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[Serusers] Unable to complete calls with 0.8.13
by Andres 30 Jan '04

30 Jan '04

Hi, I downloaded the latest CVS tarball and installed it in our lab. Using a LAN setup, where SER and 2 Sipura units are all in the same subnet, I am unable to complete any calls. All I can tell is that SER is unable to match the ACK to a current transaction. So when phone A makes a call to phone B and it answers, phone B sends the STATUS 200 OK back. Phone A then sends the ACK, but SER is unable to match it and is not forwarded to phone B. So the call is not completed. The syslog just shows a bunch of these messages: Warning: sl_send_reply: I won't send a reply for ACK!! I have tried this under RedHat 7.3 and Solaris 8 with the exact same results. I can send the Ethereal Trace of this event to whoever can help me. The config file is extremely simple (listed below). Any clues of what is going on? Thanks. -- Andres Network Admin http://www.telesip.net 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") { 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(); }; } -- Andres Network Admin http://www.telesip.net

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[Serusers] failure route and ERROR: t_should_relay: status rewrite by UAS: stored: 408, received: 487
by Alan Crosswell 29 Jan '04

29 Jan '04

I'm trying to do failure route to voicemail (which is working) but this error is logged: ERROR: t_should_relay: status rewrite by UAS: stored: 408, received: 487 I googled this and see that this came up last in October (http://lists.iptel.org/pipermail/serusers/2003-October/002921.html) but I don't see any evidence of a solution in the thread. I suspect mine is the same problem as I have two UAs registered so the initial dset is two places and then the append_branch on timeout is just the voicemail uri. BTW, I am doing this with asterisk for the usual DTMF access reasons (altough I haven't yet figured out how to map sip:alan@columbia.edu to DTMF. Maybe I'll have to give up and do numeric mailboxes:-( I am rewriting with a prefix of "vm*u" before punting over to asterisk which has exten => _vm*u.,1,Wait,1 exten => _vm*u.,2,Voicemail(${EXTEN:3}) exten => _vm*u.,3,Goto(#,1) Any pointers would be appreciated. My not-yet-complete ser.cfg is attached (not yet punting voicemail for unregistered subscribers; just registered subscribers who time out the invite). /a # # $Id: ser.cfg,v 1.21.2.1 2003/07/30 16:46:18 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=2 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) #listen=128.59.39.127 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" alias="columbia.edu" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/acc.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/exec.so" # ----------------- setting module-specific parameters --------------- # -- transaction timers -- modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "fr_timer", 10 ) # -- usrloc params -- # modparam("usrloc", "db_mode", 2) modparam("usrloc", "timer_interval", 10) # -- auth params -- modparam("auth_db", "calculate_ha1", yes) 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) # -- acc params -- modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1 ) modparam("acc", "log_missed_flag", 2) # ------------------------- request routing logic ------------------- # main routing logic route{ /* ********* ROUTINE CHECKS ********************************** */ # 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; }; lookup("aliases"); # 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; }; setflag(2); # 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=~"^sip:(.+@)?columbia.edu") { if (method=="REGISTER") { log(1, "REGISTER received\n"); if (!www_authorize("columbia.edu", "subscriber")) { www_challenge("columbia.edu", "0"); break; }; save("location"); break; }; /* ********** Dial out to PSTN logic ************* */ # 5 Digit dialing, interior calls if (uri=~"^sip:[1347][0-9]{4}@columbia\.edu") { rewritehostport("128.59.59.242:5060"); log(1,"5 digit expression match"); route(2); break; }; # 10 Digit dialing with outlide line if (uri=~"^sip:931[0-9]{10}@columbia\.edu") { if(!(src_ip=="128.59.59.242") & !(proxy_authorize("columbia.edu","subscriber"))) { proxy_challenge("columbia.edu", "1"); break; } else { rewritehostport("128.59.59.242:5060"); log(1," 93 Outside line with 10 digit expression match"); route(2); break; }; }; /* voicemail access */ if (uri=~"^sip:\*86@columbia\.edu" |uri=~"^sip:vm@columbia\.edu" |uri=~"^sip:voicemail@columbia\.edu") { route(3); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { if (!exec_dset("/etc/ser/sipldap")) { sl_send_reply("404", "Not Found"); break; } else { log(1," sipldap call"); }; }; #!lookup }; if (method == "INVITE") { t_on_failure("1"); }; # 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(); }; } route[2] { log(1,"route[2]:SIP-to-PSTN call routed"); if (!t_relay()) { sl_reply_error(); }; } # ---- voicemail user access ---- route[3] { rewritehostport("127.0.0.1:5069"); log(1,"voicemail access"); if (!t_relay()) { sl_reply_error(); }; } # ------------- handling of unregistered user ------------------ route[4] { log(1,"route[4]: user not registered"); # non-Voip -- just send "off-line" if (!(method == "INVITE" || method == "ACK" || method == "CANCEL")) { sl_send_reply("404", "Not Found"); break; }; # not voicemail subscriber # if (!isflagset(4)) { # sl_send_reply("404", "Not Found and no voicemail turned on"); # break; # }; # forward to voicemail now prefix("vm*u"); rewritehostport("127.0.0.1:5069"); t_relay_to_udp("127.0.0.1", "5069"); } failure_route[1] { # transfer to asterisk voicemail with uMAILBOX for unavailable. # sip:USER@columbia.edu -> sip:vm*uUSER@127.0.0.1:5069 t_on_failure("2"); prefix("vm*u"); rewritehostport("127.0.0.1:5069"); append_branch(); log(1,"redirection to voicemail\n"); t_relay(); } failure_route[2] { # forwarding failed (voicemail down?) log(1,"voicemail failed\n"); t_reply("500","Weasels have eaten voicemail again"); }

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sr-users January 2004