Hello,
I've been browsing the net and the (not so friendly) documentation but found no readily available solution. I have tried to fabricate on myself but it seems it's not that easy without getting deeply familarised with SIP protocol internals which I don't quite want know.
The problem seems simple enough, maybe you can easily tell the solution.
There is kamailio at ip 256.10.10.1. It's task is to accept connections from SIP clients and forward them to PBX at 256.20.20.1, and accept PBX packets and forward them to the clients. PBX handles registration, options, invites, everything. Kamailio does not authenticate, authorise, just forwards everything back and forth, like a mindless proxy. No NAT involved.
[The reason is that K. have to call external scripts whenever an outbound call (towards the clients) is initiated, which the PBX is unable to. Please don't advise other solutions (which isn't able to call executables in-session).]
I have created a simplistic config which is not expected to work and it doesn't quite (plenty of security stuff left out now):
route { if(src_ip==myself) { add_path(); $du="sip:256.20.20.1:95060"; } else { if(src_ip!="256.20.20.1") { $du="sip:256.20.20.1:95060"; } else { #from gateway pkts aren't changed }
if(!t_relay()) { sl_reply_error(); break; } }
There is several possible problems, but it seems Contact: is one main one since it contains the client instead of the proxy. Possibly this may even be the almost-good solution.
I'm sure there's an easy and pretty solution so before I start to make ugly ones I ask the wise people.
Thanks, Peter
Hi Peter,
I created simple config for freeswitch+kamailio set-up, where everything including registration is handled by Freeswitch(PBX) and you can try to use it, maybe this is what you need.
https://github.com/os11k/dispatcher
I never tested in in prod, so use it on your own risk.
With kind regards,
Jurijs
2015-12-17 17:36 GMT+02:00 Peter Gervai grinapo@gmail.com:
Hello,
I've been browsing the net and the (not so friendly) documentation but found no readily available solution. I have tried to fabricate on myself but it seems it's not that easy without getting deeply familarised with SIP protocol internals which I don't quite want know.
The problem seems simple enough, maybe you can easily tell the solution.
There is kamailio at ip 256.10.10.1. It's task is to accept connections from SIP clients and forward them to PBX at 256.20.20.1, and accept PBX packets and forward them to the clients. PBX handles registration, options, invites, everything. Kamailio does not authenticate, authorise, just forwards everything back and forth, like a mindless proxy. No NAT involved.
[The reason is that K. have to call external scripts whenever an outbound call (towards the clients) is initiated, which the PBX is unable to. Please don't advise other solutions (which isn't able to call executables in-session).]
I have created a simplistic config which is not expected to work and it doesn't quite (plenty of security stuff left out now):
route { if(src_ip==myself) { add_path(); $du="sip:256.20.20.1:95060"; } else { if(src_ip!="256.20.20.1") { $du="sip:256.20.20.1:95060"; } else { #from gateway pkts aren't changed }
if(!t_relay()) { sl_reply_error(); break; } }
There is several possible problems, but it seems Contact: is one main one since it contains the client instead of the proxy. Possibly this may even be the almost-good solution.
I'm sure there's an easy and pretty solution so before I start to make ugly ones I ask the wise people.
Thanks, Peter
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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Jurijs Ivolga -
Peter Gervai