[Asterisk-Users] RE: Setup/SIP routing

[E. Versaevel]

The problem is that that should be dynamic :/

Take a look at this sip msg:

	INVITE sip:erik at localphone:5061 SIP/2.0
	Max-Forwards: 10
	Record-Route: <sip:3400009521 at 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 at ser.box:5065>;tag=as3f718642
	To: <sip:3400009521 at sip.infopact.com>
	Contact: <sip:asterisk at ser.box:5065>
	Call-ID: 533cb84a48058ebb71fbd7bf7557c0f0 at 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 at 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 at 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 at 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 at 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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