--- Klaus Darilion <[EMAIL PROTECTED]> 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:[EMAIL PROTECTED] > ip > > > > When agent send INVITE to an other agent ASTERISK > use > > > > AOR sip:[EMAIL PROTECTED] 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 <[EMAIL PROTECTED]> > 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 === ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users