sr-users October 2008

sr-users@lists.kamailio.org

93 participants
169 discussions

[Kamailio-Users] Send MaxCallDuration and UserCredit for freeradius-cdrtool

by Diego Zuaneti Arruda

Hi all, How I send to freeradius-cdrtool the maxCallDuration and UserCredit? On the /etc/raddb/users I have: 87001(a)mydomain.com.br Auth-Type := Digest, User-Password == "mypassword" Reply-Message = "Authenticated", SIP-AVP = "authentic#1", SIP-AVP = "callMaxDur:60", SIP-AVP = "credit:25" Is this correct ? Thanks. Diego.

15 years, 7 months

Re: [Kamailio-Users] How to handle DoS attack and OpenSER Pikemodule!

by arif.zaman

arif.zaman wrote: > > Hello Daniel, > > I've tested according to your suggessions but No Luck. I've configured to > block for 10 minutes and its working but not works for 2 hours. > >> In my case: If the number of SIP messages from a single IP address to >> my SIP Proxy exceeds 200 per minute. Recommended action: Block IP >> for 2 hours. >> > >> modparam("pike", "sampling_time_unit", 60) modparam("pike", >> "reqs_density_per_unit", 200) modparam("pike", "remove_latency", 7200) > Is there any help? Thanks, ARIF -- View this message in context: http://www.nabble.com/How-to-handle-DoS-attack-and-OpenSER-Pike-module%21-t… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

15 years, 7 months

[Kamailio-Users] lcr gateway can now be given a hostname

by jh＠tutpro.com

lcr gateway can now be given a hostname that is used in request-uri when request is sent to the gateway. hostname is not, however, used in forwarding the request. forwarding is still based on gateway's ip address, port, and transport protocol. this is in order to be able to keep on supporting from_gw() and to_gw() functions. -- juha

15 years, 7 months

[Kamailio-Users] Check if a user is calling from a registered device and if not, deny the call!

by Arif-Uz-Zaman

Hi, I do like to configure OpenSER in such a way so that it can accept incoming call from a registered user only. After receiving SIP INVITE message, OpenSER will check whether caller is registered or not before forwarding that received INVITE message to make sure only registered user will make call through OpenSER. If Caller is not registered, OpenSER will deny the call. Is it possible to check if a user is calling from a registered device without using any Database? Please help me with sample configuration by considering my interest. Thanks, ARIF

15 years, 7 months

[Kamailio-Users] Check if a user is calling from a registered device and if not, deny the call!

by jh＠tutpro.com

Arif-Uz-Zaman writes: > OpenSER will check whether caller is registered or not before forwarding > that received INVITE message to make sure only registered user will make > call through OpenSER. If Caller is not registered, OpenSER will deny the > call. > Is it possible to check if a user is calling from a registered device > without using any Database? Please help me with sample configuration by > considering my interest. arif, i'm quite sure that this same question has come up many times before already. please check the mailing list archives. -- juha

15 years, 7 months

[Kamailio-Users] OpenSER 1.3.2 + Mediaproxy 1.9.2 + Asterisk 1.4 + CDRTool with FreeRadius

by Nuno Marques

Hi to all, I need to implement a simple prototype of a Voip system with OpenSER 1.3.2 + Mediaproxy (for NAT traversal) + Asterisk 1.4 (for interconnection with PSTN) + CDRTool (for accounting) on a debian 4 etch operating system. I've already configured OpenSER + Mediaproxy + Asterisk with sucess. I'm missing CDRTool. I have two questions, if some one can help me: 1º which is the CDRTool version compatible with openser 1.3.2 and mediaproxy 1.9.2. Can i use the latest (6.6.10 i supose)? 2º I've read the install file for CDRTool but it is a little confusing (maybe because i'm a newbie to all this). Can anyone show me some literature (or tutorials) to accomplish this installation? Thanks in advance to all, Nuno

15 years, 7 months

[Kamailio-Users] simple question

by Alex R.S.M

Hi, I want openSER to act ad a B2B UA, replacing TO, FROM and URI fields. I know "rewritehostport" function allow me to modify URI. What are functions can be used to change TO and FROM fields. Thanks, _________________________________________________________________ Get more out of the Web. Learn 10 hidden secrets of Windows Live. http://windowslive.com/connect/post/jamiethomson.spaces.live.com-Blog-cns!5…

15 years, 7 months

[Serusers] About a SIP load balancing document involving SER

by Iñaki Baz Castillo

Hi, I've found a document about SIP load balancing in iptel.org website and I would like to comment an error I've found in the document. The document is this: "Towards Effective SIP load Balancing" http://www.iptel.org/voipsecurity/doc/14%20-%20Kambourakis%20-%20Towards%20… Page 7 says: -------------------- SIP Proxies insert a VIA header in all the incoming SIP requests So… In case the LB (Load Balancer) is implemented as another proxy all SIP responses will pass through that proxy Request that belong to a specific dialog should not pass through LB -------------------- This is completely true, but next page (8) says: -------------------- Transparency for responses Prevent Load Balancer from inserting a VIA header E.g. in SER utilizing the SEND core command Modify the SIP's Proxy core to ignore the VIA-header added by the Load Balancer ------------------- There is an important error that unfortunatelly I've realized it's very common. Section 18.2.2 of RFC 3261 says clearly that the responses are *always* sent through the same nodes the request came from. So the response should always traverse the load balancer. 1) Load balancer --- (SIP UDP) ---> UAS In this case the UAS would always reply to the *real* source IP (if this is different of the Via "sent-by" then UAS adds "received" parameter and replies there). 2) Load balancer --- (SIP TCP/SCTP) ---> UAS By definition a UAS must reply using the incoming TCP connection. So it's extrange for me that a document about SIP load balancing tries to offer solutions that are not SIP compliant and also unfeasible (UAS will always reply to the real source IP regardless of the Via content). Since the document proposes SER based solutions (using "SEND" command) I'd just like to confirm if I'm completely right, or maybe it's common those not SIP compliant methods by some vendors in order to provide a load balancing solution. Thanks a lot for any comment and best regards. -- Iñaki Baz Castillo

15 years, 7 months

Re: [Kamailio-Users] In-dialog request correlation without loose routing?

by Alex Balashov

Klaus Darilion wrote: > So do you perform lookup() also for in_dialog requests? When necessary. Otherwise, A endpoint just provides the URI of the Z-end of the signaling path for the domain and I make special exceptions to relay that as long as I have the Call-ID stored somewhere, which I do. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

15 years, 7 months

Re: [Kamailio-Users] In-dialog request correlation without loose routing?

by Alex Balashov

Klaus Darilion wrote: > Alex Balashov schrieb: >> BTW, I do think it would be a good idea for the dialog module to >> export these functions directly into the script symbols so they can be >> called that way. I do not like to do loose routing unnecessarily / >> when I have no use for it. > > How does your setup work without loose_route? The proxy sees in-dialog > requests only if you record_route. If you do record_route(), you have to > use loose_route for correct routing. You can't have one without the > other (except you do manual routing also for in-dialog requests) That's generally what I always do - manual routing for in-dialog requests. I was not aware that loose_route() is required for correct routing of in-dialog requests when they are record-routed back through the proxy. I think one of the reasons why this may not be much of an issue for me is because my proxy applications generally always have the proxy as the URI domain - I won't relay for !uri == myself. So, after the initial INVITE is rewritten, the UAC/UAS cores on either side send subsequent in-dialog requests to the same URI they sent the INVITE to. I would, of course, be eager to hear any methodological insights you may offer about what I'm doing incorrectly. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

15 years, 7 months

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sr-users October 2008