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:[EMAIL PROTECTED],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:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK6fe1dcea
From: "My ATA" <sip:[EMAIL PROTECTED]>;tag=as0ff4afbb
To: <sip:[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username="41911234567", realm="asterisk", algorithm="MD5",
uri="sip:[EMAIL PROTECTED]", 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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http://www.ticinocom.com
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