asterisk users - Nov 2004 - Setup/SIP routing

Hello,

I'm still kinda new to asterisk, but I'm trying to setup the following
situation:

Aterisk running at port 5065, SER running at 5060 (done that, works fine)

Load of SIP clients registering at SER (no problem), SER routing the SIP
traffic to Asterisk (no problem).

Incomming:
*) Asterisk registers at a SIP provider which sends SIP traffic to my host,
but not addressed to the registered user.
*) Asterisk has to pickup that sip traffic and relay it to ser at the same
host, but should not mess up the addressed user.
*) Ser performs routing (no problem here)

Outgoing:
*) SER sends outbound traffic to Asterisk (no problem)
*) Asterisk has to dial out to that number, but is challenged by my SIP
provider, asterisk answers challenge (should work by setting up a peer)

The reason for this setup is that my SIP Provider requires me to register at
his proxy (SER can't do that) and has to authenticate outbound calls (SER
can't do that either)

However, I'm having troubles routing incoming sip traffic to SER, asterisks
keeps messing up the form header (replacing it by the dialed context, ie
s@serhost )

Any ideas if this setup is even possible?

Kind regards, 

E. Versaevel

The problem is that that should be dynamic :/

Take a look at this sip msg:

	INVITE sip:erik@localphone:5061 SIP/2.0
	Max-Forwards: 10
	Record-Route: <sip:3400009521@ser.box;ftag=as3f718642;lr=on>
	Via: SIP/2.0/UDP ser.box;branch=z9hG4bK93dc.71ad80b3.0
	Via: SIP/2.0/UDP ser.box:5065;branch=z9hG4bK513b584d
	From: "3400009525" <sip:asterisk@ser.box:5065>;tag=as3f718642
	To: <sip:3400009521@sip.infopact.com>
	Contact: <sip:asterisk@ser.box:5065>
	Call-ID: 533cb84a48058ebb71fbd7bf7557c0f0@ser.box
	CSeq: 102 INVITE
	User-Agent: Asterisk PBX
	Date: Thu, 18 Nov 2004 12:46:51 GMT
	Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
	Content-Type: application/sdp
	Content-Length: 220
	P-hint: USRLOC

	v=0
	o=root 26383 26383 IN IP4 ser.box
	s=session
	c=IN IP4 ser.box
	t=0 0
	m=audio 14682 RTP/AVP 0 101
	a=rtpmap:0 PCMU/8000
	a=rtpmap:101 telephone-event/8000
	a=fmtp:101 0-16
	a=silenceSupp:off - - - -

As you can see the from user is not correct, this should be
3400009525@domain. If a user adds this entry to a phonebook, the contact
info will be wrong.




-----Oorspronkelijk bericht-----
Van: Benjamin on Asterisk Mailing Lists
[mailto:benjk.on.asterisk.ml@gmail.com] 
Verzonden: donderdag 18 november 2004 11:41
Aan: E. Versaevel
Onderwerp: Re: [Asterisk-Users] Setup/SIP routing

Hi

On Thu, 18 Nov 2004 11:32:08 +0100, E. Versaevel <erik@infopact.nl>
wrote:
> However, I'm having troubles routing incoming sip traffic to SER,
asterisks
> keeps messing up the form header (replacing it by the dialed context, ie
> s@serhost )
You can control what Asterisk puts into the FROM header through the
parameters "fromuser" and "fromdomain" in sip.conf.

regards
benjamin
-- 
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.

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