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 klaus.mailinglists@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+serprivate net=====nathelper ======nat===private net
nat box || internet======I suggest:
- 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
- Setup the firewall (e.g. iptables) correctly to
allow traffic from/to ser, asterisk and the RTP proxy
Done
- 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
- 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.
- Do really read every word in the "getting
started" document, if things are unclear read it again.
- Do not post "how to make this setup". Ask small
questions addressing particular (small) problems.
- 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).
always remember that this support is voluntary
If you don't find the proper english word, look
into the dictionary instead of using another word which might also have other meanings.
- Go and buy an english SIP book. (this will you
help to learn the english terms for all the SIP stuff)
- use ngrep to watch the SIP call flow
# ngrep -t -d any port 5060
regards klaus
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