[Users] open letter (2)

[Klaus Darilion]

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?

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
>>
> 
> 
> 
> 
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