[Asterisk-Users] Re: [Users] open letter (2)

[harry gaillac]

--- Klaus Darilion <klaus.mailinglists at pernau.at> a
écrit :

> Hi Harry!
> 
> As this emails are on-topic you should cc: to the
> list.
> 
> harry gaillac wrote:
> > In fact the problem is in contact  sip header
> field
> > (private ip)
> > agent send ReGISTER to SER (outbound proxy) which
> one
> > send REGISTER to ASTERISK .
> > Asterisk register agent with AOR sip:users at private
> ip
> > 
> > When agent send INVITE to an other agent ASTERISK
> use 
> > 
> > AOR sip:user at private ip but the firewall don't
> allow
> > this 
> > Asterisk SHOULD resend INVITE to SER.
> > 
> > Does SER is able to rewrite contact field in SIP
> HF?
> 
> Which IPaddress:port do you want to have in the
> REGISTER's Contact: 
> header sent from ser to Asterisk?

in fact i wish to replace all private ip in the
contact field with the public ip of ASTERISK 

Harry
> 
> klaus
> 
> > 
> > Regards
> > Thanks for your advices
> > 
> > Harry
> > 
> > 
> > --- Klaus Darilion <klaus.mailinglists at pernau.at>
> a
> > écrit :
> > 
> > 
> >>harry gaillac wrote:
> >>
> >>>>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?
> >>
> > 
> > 
> > Polycom IP300
> > 
> > 
> >>>>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.
> >>
> >>
> >>>>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).
> > 
> > 
> > SER asterisk listen on public ip
> > 
> > 
> > 
> >>2. Setup the firewall (e.g. iptables) correctly to
> >>allow traffic from/to 
> >>ser, asterisk and the RTP proxy
> > 
> > 
> > Done
> > 
> > 
> >>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.
> > 
> > 
> > Done
> > 
> >>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
> >>
> > 
> > 
> > 
> > 
> 
=== message truncated ===



	

	
		
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