sr-users May 2005

sr-users@lists.kamailio.org

230 participants
522 discussions

by michael p

hi somebody can explain me how to use sipsack command? i need to find how to do a ping in SIP to know the latency between SER and the user's phone there is something linked to sipsack (or serctl)? thanks M.

19 years, 9 months

[Serusers] help me about voicemail(sems)

by ooo ooo

when I call voicemail group user, I get error : [root@sip800 sbin]# ./sems -E Configuration: configuration file: /usr/local/etc/sems/sems.conf Ser's FIFO: /tmp/vm_ser_fifo our FIFO: /tmp/am_fifo plug-in path: /usr/local/lib/sems/plug-in/ daemon mode: 0 local IP: 192.168.0.3 (4939) WARNING: reloadModuleConfig (SemsConfiguration.cpp:66): no configuration found for module isdngw, maybe you want to specify config.isdngw in config file. (4939) ERROR: send (AmRtpPacket.cpp:119): while sending RTP packet: Invalid argu ment (4939) ERROR: send (AmRtpStream.cpp:179): while sending RTP packet.(4939) ERROR: send (AmRtpPacket.cpp:119): while sending RTP packet: Invalid argument (4939) ERROR: send (AmRtpStream.cpp:179): while sending RTP packet. who have same error , pls help me , thank you. B.R. Zhao Zigang _________________________________________________________________ 享用世界上最大的电子邮件系统― MSN Hotmail。 http://www.hotmail.com

19 years, 9 months

RE: [Serusers] Radiator + SER

by Ricardo Martinez

You have to install SER with Radius support like any other Radius Server. There is no particular configuration to work with Radiator. There is a RADIUS_HOWTO guide in iptel site... that may help you. Regards Ricardo Martinez.- > -----Mensaje original----- > De: Edgardo O. Gonzales II [mailto:edgardo.g@pacific.net.ph] > Enviado el: Miércoles, 11 de Mayo de 2005 4:01 > Para: serusers(a)lists.iptel.org > Asunto: [Serusers] Radiator + SER > > > > Hi! > > Can somebody help us on how to install SER supporting radiator. > If possible provide us a scratch for us to easily understand the > process. > > Your immediate feedback is highly appreciated. > > thanks, > ed > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

19 years, 9 months

[Serusers] socket="/tmp/ser_socket" - parse error

by davor jovanovic

Hi all, I have bad config file when I try to enable socket="/tmp/ser_socket" I'm using SER 0.8.14 from CVS. Can anyone help? Regards

19 years, 9 months

[Serusers] SER 0.9.0 on Mandrake 10.0 with 3rd Party Radius AAA

by Ozan Blotter

Hi List, Is there anyone using "Advanced VoIP Billing Software", i have a few questions regarding to it : Both Billing Software and SER installed on same machine and LAN IP is 192.168.1.11 (test system) Linux Dist : Mandrake 10.0 (OS) Apache : Installed (base web server) Java : Installed (important for tomcat) Tomcat : Installed (important for web interface) Perl & PHP : Installed (and a few perl modules for various applications) MySql : Installed (advanced voip needs it and creates to database like "raddb" and "voipbilling") PostgreSql : Installed (no need for it at this time) Radiusclient-ng : Installed (it's needed for SER itself) I followed steps on http://www.iptel.org/ser/doc/ser_radius/ser_radius.html except freeradius server as normal. I've installed SER from RPM's for Mandrake: ser-0.9.0-1mdk.i586.rpm ser-jabber-0.9.0-1mdk.i586.rpm ser-mysql-0.9.0-1mdk.i586.rpm ser-radius-0.9.0-1mdk.i586.rpm <<<<< SER.CFG >>>>> [root@localhost ser]# cat ser.cfg # # $Id: ser.cfg,v 1.27 2005/03/10 14:16:25 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) #memlog=5 # memory debug log level #log_facility=LOG_LOCAL0 # sets the facility used for logging (see syslog(3)) /* 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" #user=ser #group=ser #fifo_user=ser # owner of the ser fifo #fifo_group=ser #fifo_mode=0660 # fifo's permissions #disable_core=yes #disables core dumping #open_fd_limit=1024 # sets the open file descriptors limit #mhomed=yes # usefull for multihomed hosts, small performance penalty #disable_tcp=yes #tcp_accept_aliases=yes # accepts the tcp alias via option (see NEWS) # ------------------ module loading ---------------------------------- 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" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/acc.so" loadmodule "/usr/lib/ser/modules/auth_radius.so" loadmodule "/usr/lib/ser/modules/xlog.so" # ----------------- setting module-specific parameters --------------- modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("auth_radius", "service_type", 15) 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 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 (!radius_www_authorize("")) { www_challenge("","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(); }; } <<<<< SER --- When I run it >>>>> [root@localhost root]# ser -c 0(2178) WARNING: fix_socket_list: could not rev. resolve 192.168.1.11 0(2178) WARNING: fix_socket_list: could not rev. resolve 192.168.1.11 Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 192.168.1.11 [192.168.1.11]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 192.168.1.11 [192.168.1.11]:5060 Aliases: tcp: localhost:5060 udp: localhost:5060 config file ok, exiting... [root@localhost root] <<<<< RADIUS --- When I run it >>>>> [root@localhost advradius]# ./runserver.sh Wed, 11 May 2005 13:05:58 095: FINE: <GlobalConf> AuthenticationPort = 1812 AccountingPort = 1813 BindAddress = null DictionaryFile = /billing/advradius/conf/dictionary LogDirectory = /billing/advradius/log LogLevel = FINE ConsoleWrite = true FileWrite = true ConsiderSrcIPAsNASIP = false </GlobalConf> Wed, 11 May 2005 13:05:58 739: FINE: <ThreadPoolManager> NumThreads = 10 MaximumWorkTime = 5000 KeepAliveTime = -1 ExpiryThreadSleepInterval = 5000 </ThreadPoolManager> Wed, 11 May 2005 13:05:58 882: FINE: <Client DEFAULT> NASIdentifier = default Secret = mysecret Realm = db </Client> Wed, 11 May 2005 13:05:58 917: FINE: <Realm db> 18AuthBy = [db] AcctPolicy = BACK-UP Accounting = [db, file_acct] Session = db </Realm> Wed, 11 May 2005 13:05:59 708: FINE: <DBParams DEFAULT> DriverName = org.gjt.mm.mysql.Driver DBURL = jdbc:mysql://127.0.0.1:3306/voipbilling UserName = root UserPassword = </DBParams> Wed, 11 May 2005 13:05:59 967: FINE: <AuthBy db> CheckList = [(5, (260, 0)) , (6, (0, 0)) ] ReplyList = [(1, (101, 9)) , (2, (102, 9)) , (3, (103, 9)) , (4, (6, 0)) , (7, (0, 0)) ] BackStore = DB PwdEncType = 0 SimultaneousLogin = 0 BackStore = DB DBParams = null AuthQuery = Select CallAmnt, CallTime, ReturnCode, AuthServiceType, Password, OtherCheckItems, OtherReplyItems from TblActiveSessions where AuthConfID = $cisco-h323-conf-id and AuthType = @GetRequestNumber@ AttribMapping = ColMapped </AuthBy> Wed, 11 May 2005 13:05:59 990: FINE: <Session db> BackStore = DB InsertSession = Insert INTO TblActiveSessions ( UserName, TimeStart, AuthType, CalledNumber, AuthCallingNumber, AuthNASIPAddress, AuthNASPort, AuthNASPortType, AuthConfID, CallOrigin, CallType, AcctSessionID ) values ($User-Name, $Event-Time, 3, $Called-Station-Id, $Calling-Station-Id, $NAS-IP-Address, $NAS-Port, $NAS-Port-Type, $cisco-h323-conf-id, $cisco-h323-call-origin, $cisco-h323-call-type, $Acct-Session-Id) DeleteSession = Delete from TblActiveSessions where AuthConfID = $cisco-h323-conf-id CountSession = Select count(*) from TblActiveSessions where LoginName = $User-Name and AuthConfId=$cisco-h323-conf-id BackStore = DB </Session> Wed, 11 May 2005 13:06:00 039: FINE: <Accounting db> BackStore = DB InsertAcct = Insert INTO TblCallsRad ( TimeClose, UserName, ActualDuration, CallingNumber, CalledNumber, ConfID, CallOrigin, CallType, AcctSessionID, NASIPAddress, BytesIn, BytesOut, DelayTime, AcctStatusType, RemoteAddress, RemoteGatewayID, TerminationCause, ) values ( $Event-Time, $User-Name, $Acct-Session-Time:NUMBER, $Calling-Station-Id, $Called-Station-Id, $cisco-h323-conf-id, $cisco-h323-call-origin, $cisco-h323-call-type, $Acct-Session-Id, $NAS-IP-Address, $Acct-Input-Octets:NUMBER, $Acct-Output-Octets:NUMBER, $Acct-Delay-Time:NUMBER, $Acct-Status-Type, $cisco-h323-remote-address, $cisco-h323-gw-id, $cisco-h323-disconnect-cause, ) QueueLength = 5000 RecordAcctStart = FALSE DBParams = null InsertCheckpointAcct = null </Accounting> Wed, 11 May 2005 13:06:00 070: FINE: <Accounting file_acct> BackStore = FILE InsertAcct = null QueueLength = 5000 RecordAcctStart = TRUE FileNameFormat = yyyy-MM-dd-HH.csv AcctDirectory = AcctLog </Accounting> Your evaluation license will expire in 25 day(s). Wed, 11 May 2005 13:06:00 137: INFO: Listener started at: 1812 Wed, 11 May 2005 13:06:00 143: INFO: Listener started at: 1813 Started Successfully <<< Anyone have any idea how to implement this valuable software in SER because we're gonna pay for it.

19 years, 9 months

[Serusers] SER and NavisRadius not working

by davor jovanovic

Hi all, I'm trying to configure SER 0.8.14 to use authentication with NavisRadius 4.5.0. On my X-Lite (X-Lite v2.0 Build 1103m) client I'm getting "Login Failed! Contact Network Admin." I'm getting following message from NavisRadius. Why? => Reply-Message = "No check.password" I would be very grateful if someone could look at my ser.cfg and output from NavisRadius. Best regards **************************************************************************** * This is what I get on RADIUS server (different from SER machine) Client: Client-Class = "#default" Nas_Port_Normalization = off Radius_Remove_Trailing_Nul = TRUE Radius_Append_Trailing_Nul = FALSE Auto_Remove_Check_Items = TRUE Check_Authenticators = TRUE Session_Time_From_Time_Of_Day = FALSE Radius_Charset = "UTF8" Client-Class = "video" Client-Dictionary = "draft_ietf_radext_digest_auth_01" **** dictionary of radiusclient on SER machine is adjusted to dictionary of RADIUS, and RADIUS can recognize attributes - except if I MUST run sterman_aaa_sip_00 on NavisRadius???? I attached dictionary in radiusclient to this message **** Client-Secret = <hidden> Request: User-Name = "djovanov.srce" Digest-Username = "djovanov.srce" Digest-Realm = "srce.hr" Digest-Nonce = "427b5aa983df858da94c50d1f8132a69e3e703ad" Digest-URI = "sip:srce.hr" Digest-Method = "REGISTER" Digest-Response = "ca6202fe55c51501295a9bc6ab325420" Service-Type = IAPP-Register Anonymous = v0-a208-646A6F76616E6F7669632E73726365 **** RADIUS doesn't recognize code 208, which is Sip-Uri-User **** ****Am I missing Digest-Algorithm? Why SER doesn't send this attribute?**** NAS-IP-Address = 161.53.0.131 NAS-Port = 5060 Packet: Client-Name = "161.53.0.131" Packet-Type = Access-Request Packet-Identifier = 213 Packet-Length = 197 Packet-Authenticator = 2B25D6D4934B930EB21F8E1B6AEFFE50 Source-Address = 161.53.0.131 Source-Port = 33159 Destination-Address = 0.0.0.0 Destination-Port = 1812 Receipt-Time = "2005/05/06 13:44:32" Full-User-Name = "djovanov.srce" Base-User-Name = "djovanov.srce" Normalized-Nas-Port = 5060 19 <engine.worker.0> -> checkDigest[AuthHttpDigest] 19 <plugin.AuthHttpDigest.checkDigest> FAILURE -- No check.password 20 <engine.worker.0> Variable group trace Reply: Reply-Message = "No check.password" **** this is message is which I get from NavisRadius **** **************************************************************************** ************ This is my ser.cfg fifo_db_url="mysql://ser:heslo@localhost/ser" # ------------------ 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/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/auth.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_radius.so" loadmodule "/usr/local/lib/ser/modules/group_radius.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" modparam("usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("usrloc", "db_mode", 2) modparam("usrloc", "timer_interval", 10) modparam("rr", "enable_full_lr", 1) modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("auth_radius", "service_type", 15) modparam("group_radius", "use_domain", 0) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!radius_www_authorize("")) { www_challenge("", "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; }; };

19 years, 9 months

[Serusers] Ser Accepting Calls From Asterisk

by Mr Greg Plater

Hi Calls work from Asterisk to SER now. I note in Debug output: 3(1814) contact_parser(): Empty body 3(1814) parse_contact(): Error while parsing 3(1814) error: mediaproxy/getContactURI(): cannot parse Contact header The call works fine, how do I fix this message or do I leave whats working alone. Cheers Greg _____________________________________________________________________ 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, 9 months

[Serusers] VOIP analizer

by Ernesto Lucero

Hi I just finished a sip Server and i am making calls t i am trying to map the call on what it is doing so i can analize it Thanks Ernesto

19 years, 9 months

[Serusers] Early media in 183 with nathelper/ rtpproxy.

by Schweizer Laurent

Hello, I use ser 0.9.0 and I try to fork a call to a sip ipPhone and to a mobile phone via a pstn gateway. The problem is that the gateways send a 183 Session Progress with early media and if I answer with the ipPhone the nathelper do not use the SDP from the 200 OK but continues to forward the rtp stream to the gateways ( SDP received in the 183). Thanks Laurent

19 years, 9 months

[Serusers] Call Forking and Mediaproxy

by reticent

Does anyone know of a way to use mediaproxy with forked phone calls? It seems to only allocates a single media session when use_media_proxy() is called on a SIP message with multiple branches, so when one session is established and the other is torn down, the media session with media proxy is also torn down.

19 years, 9 months

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sr-users May 2005