sr-users October 2004

sr-users@lists.kamailio.org

189 participants
408 discussions

by mohammad mirzaee

HI ALL: i have the following network: cisco ATA (valid ip) ---------------------------ser+rtpproxy---------------------------quintum GW(valid ip) I want to pass media via ser+rtpproxy, ser can communicate with rtpproxy via unix socket sucessfully. The problem is when i call from ATA to quintum the voice path is one way??????????/ MY SER.CFG IS AS FOLLOWS: # # $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" loadmodule "/usr/local/lib/ser/modules/nathelper.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) { log(1, "REQUEST FOR SERVED DOMAIN------------------"); if (uri=~"^sip:3000@ 62.220.101.2"){ *********************SER VALID IP ADDRESS*********************** rewritehost("62.220.101.3");****************************QUINTUM VALID IP ADDRESS********************** force_rtp_proxy(); log(1, "-------MEDIA IS BEINGED PROXIED--------------"); forward(62.220.101.3, 5060);********************************QUINTUM VALID IP ADD************************** break; }; #we forwad to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } Appreciate any suggections mohammad

20 years, 1 month

[Serusers] one way rtyproxy

by mohammad

20 years, 1 month

[Serusers] problems about Instant Messaging and table "location"

by C Y

hello everyone: I have some problems when I configure the SIP server.I hope you can help me to solve them. The first problem: when two users communicate with Windows Messenger 5.0(SIP),they cannot send Instant Messaging.How should I configurate the SIP server to surport it? The second problem:when a user login in the SIP system,the data of field "domain"( database "ser",table "location") is NULL.It should be "voipv6.edu.cn".So I cannot find the online user through SERWEB.What should I do to solve the problem? Thanks a lot! EMAIL:cyatx1234@hotmail.com These are some informations related to my setting and problem: 1、operating system:Linux 7.0 2、SER distribution: ser-0.8.14_linux_i386.tar.gz 3、SER build: version: 0.8.14 (i386/linux) flags: STATS:Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 @(#) $Id: main.c,v 1.168.4.3 2004/06/28 15:41:21 andrei Exp $ main.c compiled on 12:28:01 Jul 27 2004 with gcc 2.95 4、SER configuration file : # # $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=yes log_stderror=no */ check_via=yes # (cmd. line: -v) dns=yes # (cmd. line: -r) rev_dns=yes # (cmd. line: -R) port=5060 #children=4 fifo="/tmp/ser_fifo" alias="voipv6.edu.cn" "210.25.130.252" "localhost" # ------------------ 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" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" loadmodule "/usr/local/lib/ser/modules/print.so" loadmodule "/usr/local/lib/ser/modules/textops.so" #loadmodule "/usr/local/lib/ser/modules/jabber.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/vm.so" # ----------------- setting module-specific parameters --------------- modparam("usrloc","db_url","sql://ser:heslo@localhost/ser") # -- 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("voipv6.edu.cn", "subscriber")) { www_challenge("voipv6.edu.cn", "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(); }; } _________________________________________________________________ 与联机的朋友进行交流，请使用 MSN Messenger: http://messenger.msn.com/cn

20 years, 1 month

[Serusers] IP phones - NAT - Internet - NAT - SER/MediaProxy - IP phones

by Bastian Schern

Hello list members, since a while I'm working with SER and if all IP phones are on the same Network everything works fine. But now I try to get the Following configuration working: +--------------+ +--------------+ | IP-Tel #96 | | IP-Tel #97 | | 192.168.0.96 | | 192.168.0.97 | +-------#------+ +------#-------+ | | +-----+ +-----+ | | +--#--#--#--#--#--+ | 192.168.0.1 | +-----------------+ | Hardware: | | NAT/Firewall | +-----------------+ | 213.191.x.x | +--------#--------+ | Internet | +--------#--------+ | 212.202.x.x | +-----------------+ | Linux: | | NAT/Firewall | | SER/MediaProxy | +-----------------+ | 192.168.1.1 | +--------#--------+ | +--------#--------+ | Switch | +--#--#--#--#--#--+ | | +-----+ +-----+ | | +-------#------+ +------#-------+ | IP-Tel #16 | | IP-Tel #17 | | 192.168.1.16 | | 192.168.1.17 | +--------------+ +--------------+ I have chosen the SER MediaProxy to solve the NAT problem with this config: --- snip --- debug=8 fork=yes log_stderror=yes check_via=no dns=no rev_dns=no port=5060 children=4 fifo="/tmp/ser_fifo" alias="universe" loadmodule "/lib/ser/modules/mysql.so" loadmodule "/lib/ser/modules/sl.so" loadmodule "/lib/ser/modules/tm.so" loadmodule "/lib/ser/modules/rr.so" loadmodule "/lib/ser/modules/maxfwd.so" loadmodule "/lib/ser/modules/usrloc.so" loadmodule "/lib/ser/modules/domain.so" loadmodule "/lib/ser/modules/uri.so" loadmodule "/lib/ser/modules/registrar.so" loadmodule "/lib/ser/modules/textops.so" loadmodule "/lib/ser/modules/mediaproxy.so" loadmodule "/lib/ser/modules/auth.so" loadmodule "/lib/ser/modules/auth_db.so" modparam("usrloc", "db_mode", 0) modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) modparam("mediaproxy", "natping_interval", 60) modparam("registrar", "nat_flag", 2) 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; }; if (method=="REGISTER") { if (uri==myself) { # Mark as NAT'ed if (client_nat_test("3")) { setflag(2); force_rport(); fix_contact(); }; if (!www_authorize("universe", "subscriber")) { www_challenge("universe", "0"); break; } else if (!check_to()) { sl_send_reply("403", "Username!=To not allowed"); break; }; if (!save("location")) { sl_reply_error(); }; } else { sl_send_reply("403", "This domain is not served here"); }; break; }; if (method=="INVITE") { if (!(is_from_local() || is_uri_host_local())) { sl_send_reply("403", "Relaying is forbidden"); break; }; t_on_failure("1"); } else if (method == "BYE" || method == "CANCEL") { end_media_session(); }; if (loose_route()) { if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; # end media session for BYE and CANCEL is done above # before entering the loose route. no need to call it here t_relay(); break; }; # Force subsequent messages to pass trough this proxy if (method == "INVITE") { record_route(); }; if (client_nat_test("3") && !search("^Record-Route:")) { # Mark as NAT'ed force_rport(); fix_contact(); }; if (method=="INVITE") { t_on_reply("1"); }; if (is_uri_host_local()) { if (!lookup("location")) { sl_send_reply("404", "User not found"); break; }; }; if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } failure_route[1] { end_media_session(); } onreply_route[1] { if (status=~"(183)|(2[0-9][0-9])") { if (client_nat_test("1")) { fix_contact(); }; use_media_proxy(); }; } --- snap --- But if I try to make a call e.g. from #97 to #16 or any other destination, I get this Message on the Phones: "Relaying is forbidden" It's also the same, if I try to call from #17 to #16 or the other Direction. What is wrong? Is there a big mistake in my config? Thanks for your support Bastian

20 years, 1 month

[Serusers] radius_is_user_in failure

by Rodrigo P. Telles

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Folks, I've been testing SER 0.8.14 Auth and ACC (radiusclient-0.4.4) in radius (freeradius 1.0.1) and its working fine. Now I tried to use group_radius module to check if a user is in a especific group and it failed. First I received a error about missing "Sip-Group" attribute and I found a mail about this error and howto correct it (http://lists.iptel.org/pipermail/serdev/2004-July/002339.html). Now SER doesn't return any error, just radius_is_user_in(): Failure after a long time. I have a user "1213" in group "general" on radius database like this: mysql> select * from radgroupcheck; +----+-----------+-----------+----+--------+ | id | GroupName | Attribute | op | Value | +----+-----------+-----------+----+--------+ | 1 | general | Auth-Type | := | Digest | +----+-----------+-----------+----+--------+ mysql> select * from radcheck; +----+----------+---------------+----+-------+ | id | UserName | Attribute | op | Value | +----+----------+---------------+----+-------+ | 2 | 1213 | User-Password | == | testonly | +----+----------+---------------+----+-------+ mysql> select * from usergroup; +----+----------+-----------+ | id | UserName | GroupName | +----+----------+-----------+ | 1 | 1213 | general | +----+----------+-----------+ mysql> select * from radgroupreply; +----+-----------+---------------+----+------------+------+ | id | GroupName | Attribute | op | Value | prio | +----+-----------+---------------+----+------------+------+ | 3 | general | Reply-Message | = | Authorized | 0 | +----+-----------+---------------+----+------------+------+ I googled this but I couldn't find anything about that. Am I missing something? Thanks in advance. -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFBY1OHiLK8unYgEMQRAijVAJ4gmQ7cxUtDlWa8kaJzydxtefDULwCdFWtZ e4PSZdeQzihLBxgfJh17Tyg= =iD8l -----END PGP SIGNATURE-----

20 years, 1 month

[Serusers] Newbie question: asterisk and ser

by Ron Ramos

Hi All, Is my setup possible? Or maybe the right question is "Is this correct?" 1 test server: installed ser and asterisk(didn't really understand much of it yet). 1 CISCO 1750: with 2 FXO. 2 UA's: X-Lite Using SER only, I can make calls between extensions. Using Asterisk alone, can I also call between extensions? And is an extension on asterisk different on the exetensions at ser? How can I make an extension in SER also an extension in asterisk? Is it possible to route PSTN calls to the CISCO 1750 via asterisk? How about via ser? If both are possible, which should I use then? I've read somewhere that combining SER with Asterisk, or vice versa, will be very good, how come? Regards, Ron

20 years, 1 month

RE: [Serusers] PA module + registration handling

by Dave Bath

-----Original Message----- From: Jamey Hicks [mailto:jamey.hicks@hp.com] Sent: 15 October 2004 15:57 To: Dave Bath Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] PA module + registration handling Dave Bath wrote: > Hey guys, > > Many apologies if im just being totally dense on this one, but I've > been going back and forth for a few days now, and given how new the > presence module is there doesn't seem to be huge amounts of > documentation on it. > > I'm attempting to get the PA module for the latest (as of a few days > ago) CVS to work. All other config bits in ser seem to work fine. I > think perhaps my confusion is also about how it's supposed to work! > Perhaps if I could ask some questions, people could fill in where they > know the answers, or point me to the right place. > > I had in my config: > > if (method=="SUBSCRIBE" || method=="PUBLISH") { # BEGIN method==subscribe > > if (t_newtran()) { > > log(1,"PRESENCE: Change of state detected\n"); > > handle_subscription("registrar"); > > break; > > }; > > }; > Hi Dave, I updated modules/pa/README this morning to try to reduce confusion. There are two functions that need to be called: handle_subscription for incoming SUBSCRIBE messages and handle_publish for incoming PUBLISH. if (method=="SUBSCRIBE") { if (!t_newtran()) { log(1, "newtran error\n"); sl_reply_error(); }; handle_subscription("registrar"); break; }; if (method=="PUBLISH") { if (!t_newtran()) { log(1, "newtran error\n"); sl_reply_error(); }; handle_publish("registrar"); break; }; The function pa_handle_registration is no longer used. Presence aware UA's register in the same way as others. To update extended status on the PA, they should send PUBLISH messages to ser. Jamey Thanks Jamey! This is a great help.. I was confused by all the exported functions. So do I understand that pa_handle_registration is still an exported function but its deprecated now and is not needed? If you have a moment to explain (or point me at the doc) about when the db tables get populated... I never seem to see anything in them, so would you mind giving me a few hints as to what tables I should see getting populated on which events? Many thanks for any help you can provide, Dave

20 years, 1 month

[Serusers] ser support for instant messaging and presence

by Franklin, Allen

The ser Admin's Guide claims that ser supports instant messaging and presence. Does this mean I can point my Instant Message client to ser and ser will properly forward MESSAGE messages and manage REGISTER/SUBSCRIBE/NOTIFY so that my buddy lists are properly maintiained? Do I need a special ser.cfg file for this to work? I don't see any mention of SUBSCRIBE or NOTIFY in any of the sample .cfg files in the examples directory in the ser distribution.

20 years, 1 month

[Serusers] Ser fowards invite on wrong IP Address

by Stephen

If I have Ser listening on two IP addresses it will forward a Invite from the address it received it on and not from the address the target registered at. This will cause a Nat problem as there will be no binding for that combination. In the example below Stephen is registered at xxx 162 and 888 at xxx 163 and both are behind the same Nat. A call is attempted in each direction. Is this the correct behavior? Source Destination Protocol Info 202.76.176.229 206.123.107.162 UDP Source port:5061 Destination port: 5060 202.76.176.229 206.123.107.163 SIP/SDP Request: INVITE sip:stephen@blablabla.com.au, with session description 206.123.107.163 202.76.176.229 SIP Status: 100 trying -- your call is important to us 206.123.107.163 202.76.176.229 SIP/SDP Request: INVITE sip:stephen@202.76.176.229:5061;user=phone, with session description 206.123.107.163 202.76.176.229 SIP/SDP Request: INVITE sip:stephen@202.76.176.229:5061;user=phone, with session description 206.123.107.163 202.76.176.229 SIP/SDP Request: INVITE sip:stephen@202.76.176.229:5061;user=phone, with session description 202.76.176.229 206.123.107.163 SIP Request: OPTIONS sip:sip.blablabla.com.au 206.123.107.163 202.76.176.229 SIP Status: 404 Not Found 206.123.107.163 202.76.176.229 SIP/SDP Request: INVITE sip:stephen@202.76.176.229:5061;user=phone, with session description 202.76.176.229 206.123.107.163 SIP Request: CANCEL sip:stephen@blablabla.com.au 206.123.107.163 202.76.176.229 SIP Status: 200 ok -- no more pending branches 206.123.107.163 202.76.176.229 SIP Status: 487 Request cancelled 202.76.176.229 206.123.107.163 SIP Request: ACK sip:stephen@blablabla.com.au 202.76.176.229 206.123.107.162 UDP Source port: 5061 Destination port: 5060 202.76.176.229 206.123.107.162 SIP/SDP Request: INVITE sip:888@blablabla.com.au;user=phone, with session description 206.123.107.162 202.76.176.229 SIP Status: 100 trying -- your call is important to us 206.123.107.162 202.76.176.229 SIP/SDP Request: INVITE sip:888@202.76.176.229:5060, with session description 206.123.107.162 202.76.176.229 SIP/SDP Request: INVITE sip:888@202.76.176.229:5060, with session description 206.123.107.162 202.76.176.229 SIP/SDP Request: INVITE sip:888@202.76.176.229:5060, with session description 206.123.107.162 202.76.176.229 SIP/SDP Request: INVITE sip:888@202.76.176.229:5060, with session description 202.76.176.229 206.123.107.162 SIP Request: CANCEL sip:888@blablabla.com.au;user=phone 206.123.107.162 202.76.176.229 SIP Status: 200 ok -- no more pending branches 206.123.107.162 202.76.176.229 SIP Status: 487 Request cancelled 202.76.176.229 206.123.107.162 SIP Request: ACK sip:888@blablabla.com.au;user=phone

20 years, 1 month

Re: [Serusers] memory and cpu usage

by Marian Dumitru

Hi Evgeny, SER does accounting - generates CDR records, but it doesn't do billing. For this you need a billing application. To avoid loosing BYEs, you have to take care from your script by using record_routing. Best regards, Marian Dumitru Evgeny Ivanov wrote: > While I was reading the documentation I understood that SER is > accountig calls only on begining and end ot transaction - on INVITE > and BYE. So how the billing is done ? > What if SER does not receives a BYE signal ? > > > > On Mon, 11 Oct 2004 15:37:25 +0200, Marian Dumitru > <marian.dumitru(a)voice-sistem.ro> wrote: > >>Hi Pavel, >> >>SER is used only when the call is establish and terminated. During the >>actual call, there is no usage of SER server, so basically you can have >>as many user talking in parallel as you want. >> >>Anyhow, for 100 user, you don't have to worry :-) >> >>Best regards, >>Marian Dumitru >> >> >> >>Pavel Siderov wrote: >> >>>Hi, >>>Somebody can tell me what will be the cpu and memory usage and what >>>machine will I need if want for example 100 users talking at the same >>>time using the ser server? >>>Thanks in advance, >>>Pavel Siderov >>> >>> >>>------------------------------------------------------------------------ >>> >>>_______________________________________________ >>>Serusers mailing list >>>serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >> >>-- >>Voice Sistem >>http://www.voice-sistem.ro >> >>_______________________________________________ >>Serusers mailing list >>serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers >> -- Voice Sistem http://www.voice-sistem.ro

20 years, 1 month

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sr-users October 2004