sr-users February 2006

sr-users@lists.kamailio.org

189 participants
373 discussions

[Serusers] parse_msg() SIP parser test program

by Hendrik Scholz

Hi! Every once in a while I came across certain messages that apparently could not be parsed properly. To track down these I've written a small helper program. Put this into the tests/ subdirectory and follow the compilation instructions! Until now I only found STUN messages that used port 5060 so I might need to increase debugging inside the SER code. Here's the README: This program feeds SIP messages into the SER SIP parser to analyze its behaviour. Valid data sources are live traffic obtained via pcap, previously recorded traffic contained in pcap capture files or a plain containing a single SIP message. Usage: Depending on the command line parameters parser_test switches between the aforementioned modes. - monitoring live traffic: # parser_test -i <interface> -s <pcap dest> - analyzing pcap file: $ parser_test -r <pcap source> -s <pcap dest> - file mode $ parser_test -f <buffer> -s <pcap dest> needs to be set in pcap based modes. This newly generated pcap file contains all messages that could not be parsed properly by parse_msg(). Don't forget to set LD_LIBRARY_PATH to include ../lib/cds when running parser_test, i.e. $ LD_LIBRARY_PATH=../lib/cds ./parser_test -f testfile Compilation: $ gcc -c parser_test.c -o parser_test.o -Wall -g $ gcc -o parser_test parser_test.o -lcds -L../lib/cds -lpcap ../dprint.o \ ../mem/mem.o ../mem/shm_mem.o ../mem/q_malloc.o ../error.o \ ../data_lump*o ../parser/contact/parse_contact.o \ ../parser/contact/contact.o ../parser/digest/*.o ../parser/*.o -g Caveats: - Fragements are not de-fragmented. This could be done using libnids - verbose debug messages The sources as well as the README are available from: http://dump.wormulon.net/parser_test/ -- Hendrik Scholz - <hscholz(a)raisdorf.net> - http://www.wormulon.net/

18 years, 7 months

[Serusers] Help me for simple ser installation.

by Friend

Hi Friend, I am from India. Just following the Dan Austin SER HowTo and trying to install and configure ser server for the past one week no result. Feel helpless, can any one help me to make it successful. Am using ser-0.8.12-0.rpm ser-mysql-0.8.12-0.rpm RedHat Enterprise Linux 3 bind 9 MySQL 4.0.20-0 Machine: Intel x86 Can you provide me any sample files - ser.cnf, serctl, ser_mysql.sh from a successful installation. Awaiting your valuable suggestions and guidance. Sethu. --------------------------------- Yahoo! Photos NEW, now offering a quality print service from just 8p a photo.

18 years, 7 months

[Serusers] a question about "cancel" in forwarding scenario

by zhangshuai

Dear All, I want to make use of avpops module to achieve forward on-no-reply. When ua1 calls ua2 and ua2 is off-line, or busy, or doesn't answer the phone, the invite should be forwarded to ua3 and ua3 begins ringing. In my tests, first, ua1 sends "cancel" when ua2 is ringing. Ua1 receives "487 Request Terminated" from SER. Second, ua1 sends "cancel" when ua3 is ringing. So, theoretically, ua1 should receive "487 Request Terminated" after "200 canceling" from SER, right? But, ua1 received "408 Request Timeout" instead. So strange. How come? I think something is wrong with my ser.cfg. Could anyone help me about this? Thanks in advance. Here is my ser.cfg: # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/avp.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/dbtext.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" #loadmodule "/usr/local/lib/ser/modules/postgres.so" #loadmodule "/usr/local/lib/ser/modules/flatstore.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" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" loadmodule "/usr/local/lib/ser/modules/group_radius.so" loadmodule "/usr/local/lib/ser/modules/uri_radius.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", 1) # -- 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) modparam("tm", "fr_timer", 10 ) modparam("tm", "fr_inv_timer", 20) modparam("tm", "noisy_ctimer", 1) modparam("auth_radius|uri_radius|group_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("avpops","avp_url","mysql://ser:heslo@localhost/ser") modparam("avpops","avp_table","usr_preferences") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; if (!method=="REGISTER") record_route(); if (loose_route()) { t_relay(); break; }; if (method=="INVITE") { lookup("location"); #if (!lookup("location")) { # sl_send_reply("404", "Not Found"); # break; #}; } if (method=="REGISTER") { log("REGISTER\n"); if (!radius_www_authorize("localhost.localdomain")) { www_challenge("localhost.localdomain", "0"); break; }; save("location"); break; }; if (method=="INVITE") { if (!radius_proxy_authorize("localhost.localdomain")) { proxy_challenge("localhost.localdomain", "1"); break; }; } # if we do not get a positive reply, continue at failure_route[1] t_on_failure("1"); # forward the request to all destinations in destination set now t_relay(); #if (!t_relay()) { # sl_reply_error(); #}; } failure_route[1] { #when caller cancel, terminate the call if (t_check_status("487")) { t_reply("487", "Request Terminated"); break; }; if ( ( avp_db_load("$ruri","s:redirect_on_failure") && avp_check("redirect_on_failure","eq/i:1"))) { # User need to forward log(1, "User wants redirection.\n"); if ( ( avp_db_load("$ruri","s:redirectnumber") && !avp_check("redirectnumber","re/^$"))) { log(1,"first redirect\n"); #avp_print(); attr2uri("redirect number"); lookup("location"); append_branch(); #t_on_failure("2"); t_relay(); } else { t_reply("408", "TimeOut"); break; }; } else { t_reply("404", "Not Found"); break; }; }

18 years, 7 months

[Serusers] Problem with SER Accounting

by mkumar＠mantragroup.com

Hi All, I was able to install SER and call out successfully. I configured ser.cfg after seeing some examples in voip-info.org, almost all worked except accounting. I am using stable ser 0.9.4 version and followed instructions in http://www.voip-info.org/tiki-index.php?page=SER+example+accounting. When I check the database for records related tables are empty. I did not install radius, is this required to enable accounting? I am attaching my ser.cfg, please check it and guide me. Any help will be sincerely appreciated. Thanks, Manoj.

18 years, 7 months

[Serusers] changing domain

by postmaster＠orisonweb.com

hello, we had configured the mysql database and html site on hard disk.everything was working fine with domain alise sip.abcd.com for abcd.com . ip address was . later we wated to change it from sip.abcd.com to simply abcd.com . so we just removed child name sip from the domain now everything is working fine on web end but when we try to connect or log via x10 like software it prompts a error login failure contact network admin message while login in via X10. we have completely changed mysql data acordng to domain name also we made domain anme change in ser.cfg & cofin.php file but the problem is remain same if theres any ohter settings do let me know. thanks vinod bisen NOTE: Once the Domain name has been chosen it is not easy to change, since all users are recorded with the domain name, so if you change it you will have to update the config of each user's phone

18 years, 7 months

[Serusers] SER amd Minute Provider

by mkumar＠mantragroup.com

Hi All, I purchased some minutes from a minute provider and looking to configure that with SER. But I did not find any documentation or info on how to do that? Please tell me how can I do this or lead me to related documentaion or links. Thanks, Manoj.

18 years, 7 months

[Serusers] Aliases

by Nick Hoffman

Just a couple of quick questions relating to aliases :) If the environment variable SIP_DOMAIN is not set, why does adding an alias with serctl fail?: nick@customer-dev3:~> sudo serctl alias add 123456 sip:nick@altcall.com domain unknown: use usernames with domain or set default domain in SIP_DOMAIN Once I set SIP_DOMAIN like so: export SIP_DOMAIN=altcall.com Adding an alias as described above works, but why is the alias not listed in the MySQL database?: mysql> select * from aliases; Empty set (0.00 sec) Thanks and enjoy the weekend! -- Nick e: nick.hoffman(a)altcall.com p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it.

18 years, 7 months

[Users] Question on SIP authentication with users from OpenSER

by Barry Flanagan

Hi, This is more an Asterisk question, I know, but I am sure a good many of you have similar setups.... We are using Asterisk 1.2.3 with RealTime for PSTN and Voicemail where users register with an OpenSER cluster (2 nodes currently). When they request PSTN they are forwarded to * where they have entries in SIP realtime database. This ensures that they get their correct CallerID and context, etc. This is working fine at present, where I have the SIP users set up with the following relevant SIP entries: name username callerid "User" <XXXXXXX> canreinvite no context context dtmfmode RFC2833 host 123.123.123.123 insecure port type friend username username Note that I have set the host to the IP of the OpenSER server, and there is no secret. I have the OpenSER servers set up as peers also. My questions are: 1. Is this the best way to to set this up? 2. I have many users, and I need to be certain that a) the username exists and b) that the request came from one of our OpenSER servers. Will the above ensure that both the username AND the host are correct? I have seen instances where if I have a static SIP entry with the same host= line, a non-existent user will be accepted as this static user. 3. How can have more than one possible host= setting for a user (i.e. they could come in from either of our OpenSER servers. Thanks! -- -Barry Flanagan

18 years, 7 months

[Users] REMINDER: TM functions

by Bogdan-Andrei Iancu

Hi, this is reminder for the TM API clean-up. Friday 10th, the old TM functions like t_relay_to_xxx(), t_replicate_xxx(), t_forward_nonack_udp(), etc will be definitely removed. Here is the original email describing the changes: http://www.openser.org/pipermail/devel/2006-January/001659.html this changes will affect *only* the devel branch. For its users, please migrate your scripts to new function prototypes. thanks and regards, Bogdan

18 years, 7 months

[Users] sorry wrong address

by Pjothi

Sorry for the last email regarding IPSec. Inadvertently sent to the wrong group

18 years, 7 months

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sr-users February 2006