sr-users May 2005

sr-users@lists.kamailio.org

230 participants
522 discussions

by Michael Ulitskiy

Hello, I'm trying to comprehend loose routing concept and I have a question that concerns me. As far as I understand loose routing says that if there're Route headers in a message it should be forwarded according to the URIs set in Route headers. I thought that this is true only within a dialog, but RFC3261 (part 16.6) says: "Requests establishing a dialog may contain a preloaded Route header field." Also SER manual says: " the failure not to include loose routing in your scripts may lead to infinite loops. Make sure that you include the following script fragment immediately after request sanity checks" and provide the following piece of code: if (loose_route()) { t_relay(); break; }; which as far as I understand unconditionally forwards message if Route header is present. So I'm wondering what about security? If I follow this guidelines how I would shield my PSTN gateway if anyone can construct message and pre-load it with URI of my gateway and all my proxies must honor it. For example I have a PSTN gateway on ip address 10.1.1.5 and proxy on 10.1.1.10 that supposed to interface outside world. So I guess if someone construct a message like this: INVITE sip:12345@somewhere.com SIP/2.0 ... Route: <sip:12345@10.1.1.5;lr> my proxy will forward it to PSTN gateway and it will make outbound call. Is this true? Please enlighten me on this. Thank you, Michael

19 years, 1 month

[Serusers] Registrering trouble. Perhaps Fifo

by Martin Kjeldsen

Hi there I'm having trouble registrering with my SER build (0.9.2), and using the auth_mysql ser.cfg from ONsip.org. It is as if SER isn't recieving any of the messages from my UA, and I've been wondering where I might have gone wrong. One thing I've found different from a former build (0.8.14), I have up and running is the socks which SER is listening on. When i do a serctl ps on the 0.9.2 i get this: ser@sipsite:~/running/ser0.9.0/sbin> serctl ps 200 ok 0 22531 attendant 1 22532 fifo server 2 22533 receiver child=0 sock= 212.97.222.xxx:5060 3 22534 receiver child=1 sock= 212.97.222.xxx:5060 4 22535 receiver child=2 sock= 212.97.222.xxx:5060 5 22536 receiver child=3 sock= 212.97.222.xxx:5060 6 22537 timer 7 22538 tcp receiver 8 22539 tcp receiver 9 22540 tcp receiver 10 22541 tcp receiver 11 22542 tcp main process And as far as I can see the same command on 0.8.14 yields almost the same output, except that the sock has a number like '0' and a @ sign is between scok and ip (eg: sock=0 @ ip-adress)! Can this be the cause of my troubles? I'm afraid that I'm not that much into Unix and Sockets! Any and all help will be greatly appreciated! Kind regards Martin

19 years, 1 month

[Serusers] rpid again

by Dirk Willbrandt

Hi list ! I have a Problem with the appending of an rpid with the functions avp_load_radius(), append_rpid_hf(). In my Radius DB i have an SIP-AVP field which is filled with the Value rpid:[NUMBER] When i run manuall the Radius Reply-Query i get the correct value for SIP-AVP. Now, i reconfigured my ser.cfg so that an avp_load_radius() is executed after an incoming INVITE. My Radius is answering on the command and sends the correct SIP-AVP Value for the User. Now my Problem: I can't append the RPID or anything else to the SIP-Header after the answer from the Radius. So the call is going out without the rpid in the Header. Can anyone help me ? Here is the Configure-Snip: if (method=="INVITE") { # here comes the invite if (!radius_proxy_authorize("")) { # for the digest avp_load_radius it seems that i must use an proxy auth proxy_challenge("","0"); # and an proxy challenge if(!avp_load_radius("digest")) { # here i get the SIP-AVP from the radius log(1,"Failure with the AVP-Value\n"); # if it fails i'll get a message break; # go out } avp_print(); # print all avp's (this command is never executed :( ) append_hf ("Remote-Party-ID: 111222333\r\n"); # append this silly row - it is never executed append_rpid_hf("", ";party=calling;id-type=subscriber;screen=yes"); # append the rpid from SIP-AVP - and this command is also never executed break; # go out }; route(3); # go to route block 3 break; # end } I tried also this snippet without the breaks and proxy-commands but it doesn't work. I don't know what is wrong - please help me :s Thanks ! Dirk

19 years, 1 month

RE: [Serusers] Mediaproxy in LAN (?)

by Giudice, Salvatore

I am trying to route a call from a non-routable 10.x.x.x network to a phone on the internet using ser 0.9.0 and mediaproxy 1.31. Aka: PSTN to SIP from my asterisk to an x-lite I found some references indicating that I should call: if ( src_ip = IP_GATEWAY) { force_rport(); fix_contact(); use_media_proxy(); }; Supposedly, you need the force_rport and the fix_contact to put media proxy in the middle of the two rtp sessions, but I have et to see this actually work. Has anyone been able to pass a call through media proxy from a GW on a non-routable to a routable network, or even between two ip's on the same network as the mediaproxy? If so, please share your configs. -----Original Message----- From: bogdan aux [mailto:aux1d@yahoo.com] Sent: Friday, May 13, 2005 6:16 AM To: serusers(a)lists.iptel.org Subject: [Serusers] Mediaproxy in LAN (?) I want to route RTP packets between 2 computers in the same network through Mediaproxy. I know this may have no sense, but it's just a test configuration. The call use_media_proxy() seems to have no effect, the RTP packets go directly from one computer to another. My guess is mediaproxy makes a test on the IPs (they are 192.168.147.2 and 192.168.147.3) and refuses to proxy the call. Is mediaproxy working just between different networks or it's an error in my configuration file? Yahoo! Mail Stay connected, organized, and protected. Take the tour: http://tour.mail.yahoo.com/mailtour.html _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

19 years, 1 month

[Serusers] avpops / failure route

by Martin Koenig

Hello, is it possible to use avp_write on the responses in failure route? If yes, Contact header field could be evaluated and pushed into RURI, then routing via route(x) to a new destination and finally solve the 3xx forwarding issue. Regards, Martin

19 years, 1 month

[Serusers] SER 0.9 | How to check URI against uri table ?

by lamasse＠free.fr

Hi SER Users ! Is there a mean to check that the URI submitted in 'TO' header during registration is allowed for a user in the uri table or SER(if URI and username are not the same) ?? I tried with function check_to() but it doesn't work. REGISTERs are rejected because username does not match with URI. However it's written in documentation that this function should performs a check in the uri table. Is there any other me to perform this verification ? Thanks for your help !

19 years, 1 month

[Serusers] WG: rpid again

by Dirk Willbrandt

19 years, 1 month

[Serusers] Check if user is already authenticated or registered

by Bastian Schern

Hello everybody, is it possible to check if a user is already registered or authenticated? I want to allow only registered users call to PSTN destinations. Regards Bastian

19 years, 1 month

[Serusers] SER Question

by diarmuid.oneill24＠mail.dcu.ie

Hello, I am a student in DCU University in Dublin, Ireland, and i was down loading the SIP Express Router onto a Sun Blade and i cant seem to permanently set the environment variable "SIP_DOMAIN" to my localhost. I use the followng commands in the terminal to set the variable SIP_DOMAIN to the localhost and it works but when i reboot the Sun Blade and type in "#echo ${SIP_DOMAIN}" i dont get back localhost. If you have any solutions to this problem, i would be delighted to hear of them. #SIP_DOMAIN=localhost #export SIP_DOMAIN #echo ${SIP_DOMAIN} Regards, Diarmuid.

19 years, 1 month

[Serusers] error in SQL query

by Christoph Hehl

hello. I have a Problem by login in Servweb. If i klick on the login button i get this errormessages: error in SQL query, line: 30 What's wrong? thanks

19 years, 1 month

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sr-users May 2005