The problem is that that should be dynamic :/
Take a look at this sip msg:
INVITE sip:[EMAIL PROTECTED]:5061 SIP/2.0
Max-Forwards: 10
Record-Route: <sip:[EMAIL PROTECTED];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:[EMAIL PROTECTED]:5065>;tag=as3f718642
To: <sip:[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]:5065>
Call-ID: [EMAIL PROTECTED]
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
[EMAIL PROTECTED] If a user adds this entry to a phonebook, the contact
info will be wrong.
-----Oorspronkelijk bericht-----
Van: Benjamin on Asterisk Mailing Lists
[mailto:[EMAIL PROTECTED]
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 <[EMAIL PROTECTED]> 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
> [EMAIL PROTECTED] )
You can control what Asterisk puts into the FROM header through the
parameters "fromuser" and "fromdomain" in sip.conf.
regards
benjamin
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