[Asterisk-Users] SIP Changes???

[Philipp von Klitzing]

Hi!

> The failure has just been fixed as I saw in mantis:
> http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0001738

Unfortunately that didn't solve my problem - however I am not sure 
anymore that this is related, and maybe I just have a basic 
misunderstanding concerning type=peer and type=user.

Question:
Why do I need type=peer for both cases, e.g. incoming AND outgoing calls?
I am really confused here - or someone/something else is... ;->

1. I want to be able to dial out to FWD with a Dial() statement in 
extensions.conf that does not include username or password so that these 
do not show up in the CDRs, e.g. using

  Dial(SIP/${EXTEN}@FreeWorld-out-user1)

2. The above only works if FreeWorld-out-user1 is of type=peer (and not 
type=user)

3. On an incoming FWD call * unfortunately always matches the host to the 
[FreeWorld-out-user1] section instead of the [FreeWorld-incoming] 
section, which is kind of logic becase both are peers. Then 
authentication fails because the calling user naturally doesn't have the 
correct password for FreeWorld-out-user1.

Cheers, Philipp


[FreeWorld-incoming]
context=from-FreeWorld
type=peer
host=fwd.pulver.com

[FreeWorld-out-user1]
type=peer
secret=xxxxxxxx
username=yyyyyy
fromuser=yyyyyy
host=fwd.pulver.com