asterisk users - Jun 2004 - Disable authentication on outgoing SIP calls

I am trying to make Asterisk communicate with a voice switch which doesn't
need (and like) authentication on outgoing SIP calls. I have configured it as
follows in my sip.conf:
 
[myswitch]
type=friend
host=192.168.1.100
port=5060
context=default
canreinvite=no

To dial out using this switch (it acts as a PSTN gateway) I use this in
extensions.conf:
 
exten => _0.,1,Dial(SIP/${EXTEN:1}@myswitch,90)
 
 
Incoming PSTN calls from "myswitch" work, Asterisk doesn't expect
any authentication, and doesn't get any, because the switch doesn't
support it. Outgoing calls confuse the switch, because Asterisk always wants to
authenticate something, like this:
 
Reliably Transmitting:
INVITE sip:41911234567@192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK6fe1dcea
From: "My ATA" <sip:2017@192.168.1.101>;tag=as0ff4afbb
To: <sip:41911234567@192.168.1.100>
Contact: <sip:2017@192.168.1.101>
Call-ID: 5208facd2486316b3121a2985f07e9dd@192.168.1.101
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username="41911234567",
realm="asterisk", algorithm="MD5",
uri="sip:41911234567@192.168.1.100", nonce="3135a7b3",
response="1cf43a75f985ca24a9f69ba785c2da23", opaque=""
Date: Wed, 16 Jun 2004 17:24:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 295

 
The "Proxy-Authorization" part is what I need to remove from the
INVITE request. Any clues about how I could do that? I have already browsed
Wikis and ML archives... any help is appreciated
 
Thanks
-Manuel
 


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