[Asterisk-Users] SIP peer vs. user-- how is the USER ever selected?

[Steve Murphy]

Here's a real simple question for the Asterisk Venerable and Wise Ones:




Help me understand how to name my section for the
SIP user, so there is any hope of it ever being used in my sip.conf
file.

The Wiki says that it tries to match the user name from the From: header
in the INVITE packet. If no match is found in sip.conf, it will look
thru the PEERS for a matching IP.

The From line from an INVITE looks like this, with CALLERID info
contained there (my email packages folds the single line into 3):


From: MURPHY STEVE ZZ
<sip:3337778888 at 166.69.99.229:5060;transport=udp>;isup-
oli=23;tag=SDp6an001-voip.ipprovider.net+1+1f2a0f+361a4f13

(and, of couse, the caller id info will vary from one caller to
another!)

I have tried to have a 

[ipprovider]
type=user
host=voip.ipprovider.net
...

and

[ipprovider_out]
type=peer
host=voip.ipprovider.net
md5secret=abcdefabcdefabcdefabcdefabcdefabcdef
...

But, all incoming calls go to the peer definition, as well as the
outgoing calls, and I can't authenticate just outgoing calls.

What should I rename the [ipprovider] to, so that it will be used for
incoming SIP calls?

murf



-- 
Steve Murphy <murf at e-tools.com>
Electronic Tools Company