sr-users April 2006

sr-users@lists.kamailio.org

188 participants
432 discussions

[Serusers] SER and Radiusclient-ng on 64bit Linux OS
by Alan 11 Apr '06

11 Apr '06

I've compiled and installed the sources successfully on a Sun Mircosystems 20z with Red Hat Enterprise v3 running the OS. When I try to use radius it always fails to log the user in and authenticate with radius. If I run SER on a 32 bit system with the same configurations as the 64 bit system it works fine. After reading some documentation, I caught reference in regards to radiusclient-ng only supported on 32 bit systems. Is this true? The only thing that doesn't work is when a user tries to authenticate through radius the authentication always fails on the 64 bit OS. Am I missing any pre-configuration options prior to compiling on a 64 bit OS? Please help. ~Alan Error message received from debug: pre_auth(): Credentials with given realm not found

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[Users] openser-1.0.1-rhel3.0.i386.rpm
by Nasser Dawoud /MEM/PRO 11 Apr '06

11 Apr '06

Hi everyone, I have just installed openser, openser-1.0.1-rhel3.0.i386.rpm, and I have a couple of questions: 1) is this version of OpenSer fully compliant with the Presence portion of the SIP-RFC? 2) I've noticed that presence is buggy, it does not always work correctly; that is to say, it does not represent the correct presence of the respective user in the roster. 3) I would like to use OpenSer and TLS, does that mean I only replace the libraries, *.so files, or do I have to replace everything?

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[Users] callid field in MYSQL
by Shane Burrell 11 Apr '06

11 Apr '06

What should be in this field? A unique ID or a the caller id? TIA Shane

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[Users] Random error with NAT (error in 'c field')
by Bruno Machado 11 Apr '06

11 Apr '06

Hi fellows Our system is showing an anormal behavior. Sometimes, our clients devices behind NAT cannot obtain successfuls call to other device. Take a look at the SDP headers below. These 2 dumps were made between SIP Proxy and client (at private IP 192.168.0.13). Notice that the 'c field' is differente between them. The first SDP has the public address from RTPProxy (X.Y.Z.A). This call is a example of a perfect situation. The SDP below that shows the 'c field' with a private address. When the call is established, the RTP from my phone is sent to this private IP, 192.168.0.13, in our LAN. All the sinalization is sent among the 2 devices and SIP Proxy normally. When the 'openser' process is restarted, the calls flows without any problems. Somebody knows what can be happen? Thanks in advance Bruno Machado .................. INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Type: application/sdp Content-Length: 402 P-hint: usrloc applied P-hint: nat-p2p v=0 o=11000600 8000 8002 IN IP4 192.168.0.13 s=SIP Call c=IN IP4 X.Y.Z.A..t=0 0 m=audio 56304 RTP/AVP 18 4 0 8 2 99 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 ................... INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Type: application/sdp Content-Length: 381 P-hint: usrloc applied P-hint: nat-p2p v=0 o=11000600 8000 8002 IN IP4 192.168.0.13 s=SIPCall c=IN IP4 192.168.0.13 t=0 0 m=audio 5004 RTP/AVP 18 4 0 8 2 99 101 a=sendrecv a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000..a=rtpmap:8 ....................... __________________________________________________ Faça ligações para outros computadores com o novo Yahoo! Messenger http://br.beta.messenger.yahoo.com/

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[Users] Media
by Rafael Neves Harff 11 Apr '06

11 Apr '06

Hi! I've finished installing Openser on my Slackware box. I can athenticate and make SIP calls. I need to make media traffic pass through the server. Is it possible using Openser? How? If Openser doesn't implement this feature is there a 3rd software I can use to do this using Openser? Thanks, Rafael Neves Harff from Brazil

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[Users] uuid-based setup
by Andreas Granig 11 Apr '06

11 Apr '06

Hi again, Just another question: If I'd like to switch from a username/domain-based setup to an uuid-based setup, what exactly would I have to consider? As far as I have found in the docs (or better: what I assume from what I've found in the docs), I'd have to load the uuid from subscriber like avp_db_query("select uuid from subscriber where username='$au' and domain='$ar'", "$avp(i:500)"); And then I can use i:500 as uuid for loading other avps? Or is there an implicit way to get the uuid, e.g. during www/proxy_authorize? I think I have some troubles to get the link between username/domain and uuid... Thanks, Andy

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[Users] Reading table from db
by D'Addelfio Davide 11 Apr '06

11 Apr '06

Hi all, i have configured my openser to store SIP messages into acc tables of mysql, now i need that openser read into that db's table. How can I do? Thanks

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[Users] Allow only TLS connections
by Thorsten.Haupt＠t-systems.com 11 Apr '06

11 Apr '06

Hello, I use OpenSER in a testing environment for VoIP security. My clients connect via TLS. If I deactivate UDP/5060 on the server, it doesn't work correct. Some Clients can't connect and others can't establish calls. I read in another thread, that UDP is mandatory for SIP and that the server need it. But how can I prevent users from connecting via UDP and force them to use TLS? I tried a firewall, blocking UDP and TCP on port 5060. But is this the correct way? Are there any parameters server-side to force users to connect via TLS? Thanks for response. Torsten

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[Users] OpenSER using TLS / TCP
by Thorsten.Haupt＠t-systems.com 11 Apr '06

11 Apr '06

> I think snoms are not capable of receiving incoming tls connection. And MiniSIP ... ? > Thus, you have to keep the snom-openser tls connection open ... a way to do it is to force the snom phone the register > before the tls timeout ... i think that the timeout is around 2 minutes ... Yes, this is one solution. My telephones (Snoms and MiniSIP) register now every minute. But is there no other solution, server-side? What about the parameters "tls_handshake_timeout", "tls_send_timeout" or "tcp_connect_timeout"? I tried this parameters, but nothing changed. Torsten On 4/10/06, Thorsten.Haupt(a)t-systems.com <Thorsten.Haupt(a)t-systems.com> wrote: > > > Hello, > > I have a problem with OpenSER using TLS / TCP, Clients are MiniSIP and > Snom 360 phones. > I start the server, the phones connect to it using TCP or TLS over TCP. > Calls between all phones are possible. > But after some minutes, I think it is the timeout of TLS or TCP, no > calls are possible any longer. > OpenSER shows the following error message: > > 8(31121) ERROR: tcp_blocking_connect: SO_ERROR (111) Connection > refused > 8(31121) ERROR: tcpconn_connect: tcp_blocking_connect failed > 8(31121) ERROR: tcp_send: connect failed > 8(31121) msg_send: ERROR: tcp_send failed > 8(31121) ERROR:tm:t_forward_nonack: sending request failed > > Has anyone a solution for this problem? > Thanks. > > Torsten > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

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[Users] Method BYE and loose_routed function
by buxalex＠libero.it 11 Apr '06

11 Apr '06

Hello Klaus, you were right, my BYE is loose_routed. In fact, following your suggestion, I've got the table updated. My question now is: how to know, when a call is loose_routed? I mean, in both cases now the configuration update the ACC table. should I create a if case or something like that? Thanks in advance for your help Alessandro ---------- Initial Header ----------- >From : "Klaus Darilion" klaus.mailinglists(a)pernau.at To : "buxalex(a)libero.it" buxalex(a)libero.it Cc : "users" users(a)openser.org Date : Mon, 10 Apr 2006 16:45:49 +0200 Subject : Re: [Users] new module: siptrace > Hi Alessandro! > > Please do not reply to emails when creating a new thread. Use "new > email" and choose the proper subject. Otherwise the "thread view" > feature of some email clients don't work anymore. > > Regarding your problem. May it be that the BYE is loose_routed and thus > does not hit the setflag()? > > Try setflag(2) in loose_route block. > > regards > klaus > > > buxalex(a)libero.it wrote: > > Hello to everybody, > > > > I am running openser with the module ACC and I get the db correctly updated (table ACC) for the method INVITE and ACK; But I cannot retrieve the method BYE, I am sure I have made mistake in openser.cfg, anybody can give me any help? below my openser.cfg: > > > > #----------------------------------- > > # ACC parameters > > #----------------------------------- > > > > modparam("acc", "db_url", "mysql://openser:openser@localhost/openser") > > modparam("acc", "failed_transaction_flag", 4) > > modparam("acc", "report_cancels", 1) > > modparam("acc", "db_flag", 2) > > modparam("acc", "db_missed_flag",3 ) > > .... > > > > if (method=="BYE") { > > setflag(2); > > t_relay(); > > return; > > }; > > > > > > if (method=="INVITE") { > > setflag(2); > > record_route(); > > }; > > lookup("aliases"); > > if (!uri==myself) { > > append_hf("P-hint: outbound alias\r\n"); > > route(1); > > }; > > > > thank you very much > > Alessandro > > > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > >

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sr-users April 2006