[Asterisk-Users] Configure asterisk for outgoing.. need authuser parameter?

[Tony Hoyle]

Hi,

I have access to two providers.  On one of them the authuser is the same as
the username, so outgoing works.  On the other one I can only get 
incoming -
what ever combination I try for outgoing I get an error.  The register 
command
has the ability to specify both usernames (which is why incoming works) but
outgoing doesn't seem to, and without that I'm stuck.

They are defined as:

[voiptalk]
type=peer
secret=xxxxx
username=xxxxxxx
host=voiptalk.org

[pipecall]
type=peer
secret=xxxxx
username=xxxxx
host=sipproxy.pipecall.com

The first one works OK - I can dial out with no problems.  The second one
needs an extra field for the authuser - when I try to dial out I just get:

May 17 01:03:45 NOTICE[1110916016]: chan_sip.c:5059 handle_response: 
Failed to
authenticate on INVITE to '"Tony Hoyle"
<sip:asterisk at 213.208.99.114>;tag=as4afae981'

I think this means it's using the wrong username somewhere...   I can 
dial in
just fine, so it's connected.. just only one way.

Tony

-- 
Te audire no possum. Musa sapientum fixa est in aure.

Tony Hoyle <tmh at nodomain.org>  Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724 0AA2  02B8 E547 470E 4F4B 6917