[Asterisk-Users] Disable authentication on outgoing SIP calls

[Manuel Wenger]

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 at 192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK6fe1dcea
From: "My ATA" <sip:2017 at 192.168.1.101>;tag=as0ff4afbb
To: <sip:41911234567 at 192.168.1.100>
Contact: <sip:2017 at 192.168.1.101>
Call-ID: 5208facd2486316b3121a2985f07e9dd at 192.168.1.101
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username="41911234567", realm="asterisk", algorithm="MD5", uri="sip:41911234567 at 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
 


___________________________________________________
Ticinocom SA - Via Stazione 5 - 6600 Muralto
Tel 0844 007070 - Fax 0844 007071
http://www.ticinocom.com