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?
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(a)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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