New subject: [Users] open letter (2)

23 Nov 2005

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@private ip

 When agent send INVITE to an other agent ASTERISK use 

 AOR sip:user@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 &lt;klaus.mailinglists(a)pernau.at&gt; 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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Re: [Users] open letter (2)