sr-users May 2005

sr-users@lists.kamailio.org

230 participants
522 discussions

[Serusers] Installing SER 0.9.0 from source

by Felipe Martins

Hi everybody, I'm using Slackware 10.1, and I'm migrating SER 0.8.14 to 0.9.0, is there a good howto explaining how to install it with full resources ? I found the INSTALL file (the one that comes with ser-0.9.0.tar.gz) very confusing. Thanks a lot Best Regards. -- Felipe Martins Mundivox Communications Tecnologia e Projetos fmartins(a)mundivox.com Tel.: +55 +21 +3820 8839 Cel.: +55 +21 +9823 8602 Fax.: +55 +21 +3820 8844 www.mundivox.com

19 years, 6 months

[Serusers] udp_send failed and can't get bye in accounting (Second try)

by Chris St Denis

When I try to route calls through my SER using sipp, I get there errors May 11 17:17:34 proxy-ser ser[23159]: ERROR: udp_send: sendto(sock,0x283555b8,650,0,0x28355960,16): Can't assign requested address(49) May 11 17:17:34 proxy-ser ser[23159]: msg_send: ERROR: udp_send failed May 11 17:17:34 proxy-ser ser[23159]: ERROR: t_forward_nonack: sending request failed Google didn't turn up anything interesting on them. Anyone here know whats up with them? OS: FreeBSD 5.3 SER: version: 0.8.14 (i386/freebsd) Also, I have accounting enabled to a database but I can not get it to log Bye messages. I know they are going through my server from some debug messages but they are not making it into the database. partial ser.cfg shown below. Any other suggestions on it are also appreciated. # Accounting loadmodule "/usr/local/lib/ser/modules/acc.so" # -- acc params -- # set the reporting log level modparam("acc", "log_level", 1) modparam("acc", "db_flag", 1) #modparam("acc", "db_missed_flag", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 1) modparam("acc", "early_media", 1) route{ # xlog("L_ERR", "time [%Tf] method <%rm> r-uri <%ru> - %tu\n"); # 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 -- probebly a loop or something"); 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(); /* grant Route routing if route headers present */ # 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(!(src_ip==x.x.x.x || src_ip==127.0.0.1)) { if (method=="REGISTER") { if (!www_authorize("proxy-ser.voip.aebc.com", "subscriber")) { www_challenge("proxy-ser.voip.aebc.com", "1"); break; }; save("ser_location"); break; } else if (method=="INVITE"){ if (!proxy_authorize("proxy-ser.voip.aebc.com", "subscriber")) { proxy_challenge("proxy-ser.voip.aebc.com", "1"); break; }; if(!check_from()) { sl_send_reply("403", "You can't lie about who you are"); } } } #Accounting if a call is started or ended if (method=="INVITE" || method=="BYE" || method=="CANCEL") { # xlog("L_NOTICE", "Accounting INVITE/BYE/CANCLE\n"); setflag(1); acc_db_request("", "acc"); } # native SIP destinations are handled using our USRLOC DB if (lookup("ser_location")) { xlog("L_NOTICE", "Routing - Incommming call \n"); if (!t_relay()) { sl_reply_error(); }; break; }; } else { if(!(method=="ACK")) { sl_send_reply("403", "I only relay for my own customers"); } }; # authorize only for INVITEs . RR/Contact may result in weird # things showing up in d-uri that would break our logic; our if (method=="INVITE") { ## ## Conditional relaying using if(uri=~ and rewritehostport omitted for space and security ## } # 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(); };

19 years, 6 months

[Serusers] udp_send failed and can't get bye in accounting

by Chris St Denis

When I try to route calls through my SER using sipp, I get there errors May 11 17:17:34 proxy-ser ser[23159]: ERROR: udp_send: sendto(sock,0x283555b8,650,0,0x28355960,16): Can't assign requested address(49) May 11 17:17:34 proxy-ser ser[23159]: msg_send: ERROR: udp_send failed May 11 17:17:34 proxy-ser ser[23159]: ERROR: t_forward_nonack: sending request failed Google didn't turn up anything interesting on them. Anyone here know whats up wiht them? Also, I have accounting enabled to a database but I can not get it to log Bye messages. I know they are going through my server from some debug messages but they are not making it into the database. partial ser.cfg shown below. Any other suggestions on it are also appriciated. # Accounting loadmodule "/usr/local/lib/ser/modules/acc.so" # -- acc params -- # set the reporting log level modparam("acc", "log_level", 1) modparam("acc", "db_flag", 1) #modparam("acc", "db_missed_flag", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 1) modparam("acc", "early_media", 1) route{ # xlog("L_ERR", "time [%Tf] method <%rm> r-uri <%ru> - %tu\n"); # 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 -- probebly a loop or something"); 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(); /* grant Route routing if route headers present */ # 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(!(src_ip==x.x.x.x || src_ip==127.0.0.1)) { if (method=="REGISTER") { if (!www_authorize("proxy-ser.voip.aebc.com", "subscriber")) { www_challenge("proxy-ser.voip.aebc.com", "1"); break; }; save("ser_location"); break; } else if (method=="INVITE"){ if (!proxy_authorize("proxy-ser.voip.aebc.com", "subscriber")) { proxy_challenge("proxy-ser.voip.aebc.com", "1"); break; }; if(!check_from()) { sl_send_reply("403", "You can't lie about who you are"); } } } #Accounting if a call is started or ended if (method=="INVITE" || method=="BYE" || method=="CANCEL") { # xlog("L_NOTICE", "Accounting INVITE/BYE/CANCLE\n"); setflag(1); acc_db_request("", "acc"); } # native SIP destinations are handled using our USRLOC DB if (lookup("ser_location")) { xlog("L_NOTICE", "Routing - Incommming call \n"); if (!t_relay()) { sl_reply_error(); }; break; }; } else { if(!(method=="ACK")) { sl_send_reply("403", "I only relay for my own customers"); } }; # authorize only for INVITEs . RR/Contact may result in weird # things showing up in d-uri that would break our logic; our if (method=="INVITE") { ## ## Conditional relaying using if(uri=~ and rewritehostport omitted for space and security ## } # 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(); };

19 years, 6 months

[Serusers] auth_radius

by Mihaly Zachar

Hi everybody, I'm playing with SER andd I would like to add tha auth_radius module.. versions: ser: 0.8.14 radiusclient: 0.4.8 The compilation went well, but when I start SER, it says: 0(31749) DEBUG: init_mod: auth_radius 0(31749) auth_radius - Initializing 0(31749) auth_radius: No `Cisco' vendor in Radius dictionary 0(31749) find_export: found <pre_auth> in module auth [/usr/local/lib/ser/modules/auth.so] 0(31749) find_export: found <post_auth> in module auth [/usr/local/lib/ser/modules/auth.so] 0(31749) ERROR: auth_radius: can't get code for the User-Name attribute 0(31749) init_mod(): Error while initializing module auth_radius ERROR: error while initializing modules the relevant part of my ser.cfg: ----------------------------------------- CUT -------------------------------------------- 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/exec.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_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", 2) modparam("usrloc", "db_url", "sql://xxx:yyy@localhost/ser") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") ----------------------------------------- CUT -------------------------------------------- radclient.conf: auth_order radius,local login_tries 4 login_timeout 60 nologin /etc/nologin issue /usr/local/etc/radiusclient/issue authserver localhost acctserver localhost servers /usr/local/etc/radiusclient/servers dictionary /usr/local/etc/radiusclient/dictionary dictionary /usr/local/etc/radiusclient/dictionary.sip dictionary /usr/local/etc/radiusclient/dictionary.ascend dictionary /usr/local/etc/radiusclient/dictionary.compat dictionary /usr/local/etc/radiusclient/dictionary.merit login_radius /usr/local/sbin/login.radius seqfile /var/run/radius.seq mapfile /usr/local/etc/radiusclient/port-id-map default_realm radius_timeout 10 radius_retries 3 bindaddr localhost login_local /bin/login As you can see, I have added all of the dictionary files, but still does not work :( what can be the problem???? Thanks, Regards Misi

19 years, 6 months

[Serusers] Ser Accepting Calls From Asterisk

by Mr Greg Plater

Hi Greger Thank you for the information I have found it just what I needed. Calls To SER from Asterisk: To add to trusted in MYSQL I would use this right: INSERT INTO trusted SET src_ip='***.***.***.***',proto='any',from_pattern='sip:[0-9]*@***.***.***.** *'; #replace *** with the asterisk ip address With this in my ser.cfg: modparam("permissions", "db_url", "mysql://ser:heslo@localhost/ser <mysql://ser:heslo@localhost/ser%22> ") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") and for routing to Asterisk From SER: if (method == "INVITE" && (uri=~"^sip:0")){ rewritehostport ("***.***.***.***:5060"); t_relay(); break(); }; Or do I just add: Asterisk to the trusted table and the following near the if(!allow_trusted) if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; Thank you very much for your help Cheers Greg _____ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Wednesday, 11 May 2005 6:20 PM To: Mr Greg Plater; serusers(a)lists.iptel.org Subject: Re: [Serusers] Ser Accepting Calls From Asterisk Greg, Basically, you can do four things (first two for dialing asterisk-ser, the next two for dialing ser-asterisk): 1. Accept INVITEs from a specific src ip (asterisk) and add if(src_ip != asterisk_ip) around the INVITE authentication to avoid that part being triggered 2. Or you accomplish the same thing be adding asterisk to the trusted table and us if(!allow_trusted) instead. This will give you more control as you can filter on ruri as well. 3. You can create an account in asterisk and have asterisk register with SER like a regular user (who will be available in location for dialling into asterisk). This only makes sense if you have one main number for all users registered on asterisk (you don't need 1 or 2 if you do this) 4. Use some kind of routing, either through manual configuration (uri=~"some pattern", then forward to asterisk) or use the LCR (least cost routing) module to look up the ruris and forward to asterisk. This must be done in combination with 1 or 2. Hope this helps. g-) ---- Original Message ---- From: Mr Greg Plater To: serusers(a)lists.iptel.org Sent: Wednesday, May 11, 2005 06:10 AM Subject: [Serusers] Ser Accepting Calls From Asterisk > Hi > > I am using the onsip getting started doc with mediaproxy. > > Calls for registered users on SER work great but when I try to call a > registered ser user from my asterisk box. > I get a response in Asterisk Debug. Says "failed to auth on INVITE" > > If I wish to allow calls from my asterisk box to registered users on > SER what do I change. > > > IN SER.cfg I note these 2 snippets > > > if (!www_authorize("","subscriber")) { > www_challenge("","0"); > break; > }; > > > > if (!proxy_authorize("","subscriber")) { > proxy_challenge("","0"); > break; > } else if (!check_from()) { > sl_send_reply("403", "Use From=ID"); > break; > }; > > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

19 years, 6 months

[Serusers] maxfwd.so and problems with "mf_process_maxfwd_header()" function

by reticent

I've configured call forwarding in my SER script but to catch a situation where a user forwards to another user that is forwarding back to the first user i've added a second "if ( !mf_process_maxfwd_header("10") )" stanza to the SER script. However, i'm finding that it will actually add additional "Max-Forwards" headers to the SIP message as opposed to decrimenting one that is already there: -- Max-Forwards: 9. Record-Route: *REMOVED* Max-Forwards: 10. Max-Forwards: 10. -- I am under the impression that when called the mf_process_maxfwd_header() function should search the message for the "Max-Forwards" header and only add additional headers if it is missing. tavis

19 years, 6 months

[Serusers] QoS with rtp

by Lucas Aimaretto

Hi all, I've used for long time the QoS capabilities of linux for shaping/prioritizing traffic. Now, i'm willing to prioritize rtp packets in my lan, over the rest ot packets. I've used the u32 filters, but I did not really found a nice way of "matching" a rtp packet. I read the rfc so as to look at the rtp-header and found that the only field wich does not change its value along time is the version-field. It is located at the 29th byte ( with byte 0 at the IP heder first byte ). But it is a really poor way of doing it, because there may be many other packets wich will "get caugth" by this rule ... any ideas ? any experience in the subject ... Regards, Lucas -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.8 - Release Date: 10/05/2005

19 years, 6 months

[Serusers] Good Free SipSoftphones for Linux !!!

by Felipe Martins

Hi everybody, Do anybody know a good sip softphone for linux that works with the following codecs: - G 723.1 - G 729a - G 711 - and others I'm experiencing the following problem. I have 2 ser servers, a sip softphone connected to each one of them talking to each other with no problem (softphones: phoner (M$), kphone (Linux)), my problem is that When I try to do the same test but between a softphone (linux or ms) and a HardPhone (Grandstream Budge Tone 100) I can't hear a thing. No sound is passed in any direction. I believe this is a codec problem, because between my softphones the conversation goes OK. Any Hand Thanks in Advance Best Regards. -- Felipe Martins Mundivox Communications Tecnologia e Projetos fmartins(a)mundivox.com Tel.: +55 +21 +3820 8839 Cel.: +55 +21 +9823 8602 Fax.: +55 +21 +3820 8844 www.mundivox.com

19 years, 6 months

[Serusers] (no subject)

by Juan Carlos Correa

19 years, 6 months

[Serusers] call analizer

by Ernesto Lucero

Hi Does anybody have a good call analizar for sip Thanks moyotlan

19 years, 6 months

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sr-users May 2005