harry gaillac wrote:

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Have you ever used SIP clients with presence and IM? I suggest to setup ser (without Asterisk) just to test the IM features. SIP based IM/presence implementations are very poor yet.

I've done it

And what were your experiences? Which clients do you use?

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In your picture, the NAT router is on the same PC as ser and asterisk. Is this correct?

this is correct

It would be a good idea to split things. This is a rather complicated setup.

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what scenario do you have? Are all the users behding the same NAT (in the same subnet) and you provide VoIP within this network (e.g. an enterprise) or do you have external users (e.g. like iptel or freeworlddialup)?

in fact both

            asterisk+ser

private net=====nathelper ======nat===private net nat box || internet======

I suggest:

1. Asterisk, ser and the RTP proxy 8rtpproxy or mediaproxy) should listen only on the public interface (this really must be a routable public IP address, no private).

2. Setup the firewall (e.g. iptables) correctly to allow traffic from/to ser, asterisk and the RTP proxy

3. setup ser according the "getting started" document on onsip.org. AFAIK this document contains hints how to route to a gateway. Reuse this part of the config to route certain calls to the asterisk box.

4. Try to solve things step by step: - REGISTER should work fine from Internet and LAN - Calls from Internet clients to Internet clients - Calls from LAN clients to LAN clients - Calls from LAN clients to Internet clients (and vice versa) - now try to add asterisk, e.g. calling a certain number will be routed to asterisk and starts the echo application

If all the above works (DO NOT start integrating the asterisk as long as basic SIP call do not work!!!!!), you can implement your setup.

5. Do really read every word in the "getting started" document, if things are unclear read it again.

6. Do not post "how to make this setup". Ask small questions addressing particular (small) problems.

7. Post to the related list. - do not post to developer lists - if you use ser, post to ser's list - if you use openser, post to openser's list - if you have an asterisk problem, ask at the asterisk list (e.g. you want to solve NAT traversal and registration with ser. Thus, do not ask this kind of questions at the asterisk list).

8. always remember that this support is voluntary

9. If you don't find the proper english word, look into the dictionary instead of using another word which might also have other meanings.

10. Go and buy an english SIP book. (this will you help to learn the english terms for all the SIP stuff)

11. use ngrep to watch the SIP call flow # ngrep -t -d any port 5060

regards klaus