sr-users March 2008

sr-users@lists.kamailio.org

169 participants
237 discussions

[OpenSER-Users] openser+opendiameter issue---->kindly do reply

by bhargav.b＠renovau.com

hi all, if anybody have got the information regarding authenticating a user with the help of openser and opendiameter ,pleaase help me in doing that.it seems that i am really spoiling a lot of days in doing the same by my own. i have configured openser and opendiameter correctly and even they are communicating with each other.i have used the auth_diameter module of openser.but it seems there is some problem.on the client side of the opendiameter,i am getting a messege like"invalid session id avp" when i do a call through x-lite softphone to openser. seeking ur responces regards bhargav.b

16 years, 8 months

Re: [Serusers] SIP Express Bundle. Step 1.

by Mike Trest - Personal

Hi, This is to summarize my opinions about FC* distro use. IMHO, I think FC* is best selection as it contains many more fixes than does the older CENTOS (based on 5). I have deployed several hundred FC* boxes in VoIP applications. This is over 10,000 active ports without "Enterprise" stability issues. IMHO this project needs the quickest path to the Enterprise community regardless of the OS/distro used. I suppose the ultimate question is who is our target? Ourselves, naturally. However, I suggest our target is not the bankers or major corporations with lots of rules and procedures. That group will never adopt SER until they have a commercial-grade support system to advise their IT folks what to do for every question they may have. IMHO our initial target is those early adopters who are trying to create new businesses in telecomm or consulting-on-telecom. We want them to have a solid core that they can leverage into their new appliances and specialized applications. The early adopters are risk-takers (This means us as well!) They demand an open box in which they can face the SIP world with some assurance of standards compliance while at the same time they can face their clients with something better, faster, cheaper, and innovative enough to get paid well for their efforts. Making a technology "buy - in" decision at any point in time is only a check point - not a final resting place. IMHO, we are better off selecting an OS/distro effort that has a large share of both early adopters and long term commercial support - - - so long as it meets our current and future technical **AND** target market requirements. Research confirms that the RH/FC community is the largest community with name recognition and respect among both the "geek-innovator" community as well as the Enterprise community. ..mike..

16 years, 8 months

[OpenSER-Users] Radius accounting

by Henri Keski-Sikkila

Hello, When radius accounting is in use and radius_extra is used like this: modparam("acc","radius_extra","Sip-Source-IP-Address=$si;Sip-Source-Port=$sp") and when freeradius gets an accounting-request packet, it includes: Sip-Source-IP-Address = 46.50.57.49 Sip-Source-Port = 808857653 Is this a known issue? BR, Henri Keski-Sikkila

16 years, 8 months

[OpenSER-Users] Routing calls from PSTN

by Andy Smith

Hi list, I'd like to know, from others experience, what solutions might be best for configuring OpenSER routing from a PSTN gateway to multiple IP PBXs. I guess there may be various solutions ranging from quick and dirty to the other extreme but Id be interested to get some ideas what are typical solutions for others. As OpenSER is so configurable and there are so many optional modules its quite hard, coming from zero exposure to any production SIP envrionments, to know how best to go about this... thanks Andy.

16 years, 8 months

[OpenSER-Users] Does t_realy chooseTCP/UDP protocol if it's necessary?

by Iñaki Baz Castillo

Hi, very strange: - I call to OpenSer using a TCP client. - The INVITE arrives via TCP to OpenSer. - I do: $rd=ASTERISK_IP; t_relay(); exit; - Asterisk only speaks UDP of course. - And OpenSer sends the INVITE to Asterisk by UDP. WHY does it work??? I hoped it would fail trying to forward the INVITE by TCP to Asterisk since I didn't force the outgoing socket, just the two lines above. Thanks for any explanation. -- Iñaki Baz Castillo

16 years, 8 months

[OpenSER-Users] Setting smaller register "expires" value for TCP

by Iñaki Baz Castillo

Hi, in order to mantain TCP conecction with UACs behind NAT the only working solution (AFAIK) is setting register expires value smaller than the time UAC mantains open the TCP connection (value "x"). Certainly, I don't know which is the typical "x" value (i.e. Twinkle 1.2 that suports TCP mantains TCP connection for 32 seconds). Is 30 a good value? Of course, in UDP I use "OPTIONS" keepalive from OpenSer, so I don't need (and don't want) a very small register "expires" value. I just want to set "expires" value to 30 seconds in case of TCP. But "registrar" module seems not to allow me it, I just see: default_expires (integer) min_expires (integer) max_expires (integer) It would be nice if I could set an AVP to set the max expires time in the script, and set itto 30 if "proto==TCP". Is it possible in some way? Thanks a lot for any suggestion. Regards. -- Iñaki Baz Castillo

16 years, 8 months

[OpenSER-Users] LCR and Load Balancing, Routing

by Knud Müller

Hi, I'm pretty familiar with the asterisk, but have only little understanding of SER/OpenSER. I read through some cookbooks and found some things that might help, but I just want to ensure that I'm on the right track before investing a lot of time into it. I have an asterisk running that terminates currently against one (SIP) carrier. We are noticing that the carrier has an increasing rate of short outtages and want a more redundant solution. Currently we have two more carriers and have to switch manually which was fine for the testing period. Now I'd like to route automatically. The routing decision should be based on a least cost basis, for each number the cheapest route shall be used. But if a route is not working properly, mostly these are 5XX Messages, the second cheapest route shall be used. Can I do this failover aspect with OpenSER/LCR too? Is the LCR Module the right point to start, as I have not seen failover strategies? There are some questions left which are more detailed - if a carrier shows to fail often, I just want a few calls routed to that carrier, to determine if its working again. Is there any chance to include network/packet information into the decision too, like roundtrip times, jitter etc? Hopefully someone can help me find the best solution... regards, Knud

16 years, 8 months

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sr-users March 2008