It is the way it works. First the phone sends a REGISTER without any password. Asterisk answers with a "Unauthorized" and provide a nonce to be used for the next registration attempt, using it to encrypt the password.
Leandro 2014-05-14 13:12 GMT+02:00 Olli Heiskanen <ohjelmistoarkkite...@gmail.com>: > > Hello, > > After a small break from working on this, I got the idea of tcpdumping the > correct ports. What I see is REGISTER messages from Kamailio port to > Asterisk, which are replied with 401 Unauthorized. Why is this happening? > In my sippeers table the secret field has no value (tried both NULL and > empty string) and the added field sippasswd has the correct password for > the user. > > The above might be the cause of my problem, would anyone be able to advice > me to get to correct behaviour? Now Kamailio sees the clients as > registered, which would be wrong if Asterisk doesn't. > > cheers, > Olli > > > > 2014-04-24 11:27 GMT+03:00 Olli Heiskanen <ohjelmistoarkkite...@gmail.com> > : > > >> Hello all, >> >> I've been testing a Kamailio Asterisk Realtime integration, and found a >> strange situation. >> >> My problem is that when using the integration, everything seems ok but >> Asterisk does not see the clients as registered. Kamailio and the clients >> report registered clients. Also calls fail. >> >> In Asterisk cli sip show peers shows nothing but for example realtime >> load sipusers name 660 shows the user data. Field regseconds has a value >> and fullcontact has value 'sip:660@127.0.0.1:5060' (kamailio ip:port as >> they are on the same machine). >> >> I have a very simple dialplan: >> >> [general] >> >> [default] >> exten => _XXX,1,NoOp(general : Dialed ${EXTEN}) >> same => n,Dial(SIP/${EXTEN},3600,rt) >> same => n,Hangup >> >> >> Here's more on my problem and background to it, guys on the Kamailio list >> helped out but looks like I need to check my Asterisk configuration. >> https://www.mail-archive.com/sr-users@lists.sip-router.org/msg18555.html >> >> My goal is to have all clients in the asterisk database, asterisk (one at >> this point, several later) handling the calls and Kamailio as proxy. In >> Kamailio I have the WITH_MULTIDOMAIN directive on but I'm using only one >> domain 'testers.com'. >> >> I have Asterisk 11.8.1 and Kamailio 4.2.0-dev4 on CentOS 6.5, all are on >> the same rental virtual server. Clients are in my home network behind nat. >> In MySQL I have database asterisk with table sippeers, where I have >> clients added like this: >> INSERT INTO sippeers >> (name,defaultuser,host,sippasswd,fromuser,fromdomain,callbackextension,type) >> VALUES ('660', '660', 'dynamic', 'password', '660', 'testers.com >> ','660','friend'); >> >> In this message there are some outputs and a sip trace of a register: >> https://www.mail-archive.com/sr-users@lists.sip-router.org/msg18558.html >> >> What I don't know is how to configure sip.conf, so far I've just been >> making guesses based on online examples and documentation. >> My current sip.conf looks like this: >> >> [general] >> bindport = 5070 >> bindaddr = 127.0.0.1 >> tcpbindaddr = 127.0.0.1:5070 >> tcpenable = no >> limitonpeers = yes >> ;rtcachefriends = yes >> tos_sip=cs3 >> tos_audio=ef >> realm = testers.com >> >> I've tried defining realm and domain values, but I lack proper >> understanding of those. Can you guys help me out? Are there any other >> configurations I need to check? >> >> Respectfully, >> Olli >> >> >> > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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