acl uses the remote addr from the socket connection, not anything from the sip message.<br><br><br><div class="gmail_quote">On Thu, Apr 2, 2009 at 3:07 PM, Peter P GMX <span dir="ltr"><<a href="mailto:Prometheus001@gmx.net">Prometheus001@gmx.net</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hello,<br>
<br>
I am using a SIP account from Netvoip CH. I try to receive inbound call<br>
from this SIP trunk. I discovered that, when they sent an invite, the<br>
IP-Adress of the to: is their own IP address.<br>
There fore ACL doesn't work and FS asks for authorization, which then fails<br>
<br>
I receive the following message on the CLI:<br>
2009-04-02 21:48:20 [WARNING] sofia_reg.c:1661 sofia_reg_parse_auth()<br>
Can't find user [<a href="mailto:anonymous@62.65.128.62">anonymous@62.65.128.62</a>]<br>
You must define a domain called '62.65.128.62' in your directory and add<br>
a user with the id="anonymous" attribute<br>
and you must configure your device to use the proper domain in it's<br>
authentication credentials.<br>
<br>
I could do that, but this is not clean and I do not have a password for<br>
that.<br>
<br>
How can I workaround this, so that Freeswitch accepts this call? Aliases<br>
do not seem to work.<br>
<br>
Here is a sample message after FS asks for authorization:<br>
xx.xx.xxx.xxx is the IP of our Freeswitch<br>
62.65.128.62 is the IP of Netvoip CH<br>
<br>
I would expect<br>
To: <sip:0715aaaaaa@xx.xx.xxx.xxx>.<br>
instead of<br>
To: <<a href="mailto:sip%3A0715aaaaaa@62.65.128.62">sip:0715aaaaaa@62.65.128.62</a>>.<br>
<br>
U <a href="http://62.65.128.62:5060" target="_blank">62.65.128.62:5060</a> -> xx.xx.xxx.xxx:5080<br>
INVITE sip:0715aaaaaa@xx.xx.xxx.xxx:5080 SIP/2.0.<br>
Via: SIP/2.0/UDP <a href="http://62.65.128.62:5060" target="_blank">62.65.128.62:5060</a>.<br>
Via: SIP/2.0/UDP 62.65.128.61:5060;branch=z9hG4bK8c977d2613c4d7d1fd9d03d4.<br>
Max-Forwards: 69.<br>
From: <<a href="mailto:sip%3A0049xxxxxxxxxxx@62.65.128.62">sip:0049xxxxxxxxxxx@62.65.128.62</a>>;tag=8c977d2613672832fd9d03e9.<br>
To: <<a href="mailto:sip%3A0715aaaaaa@62.65.128.62">sip:0715aaaaaa@62.65.128.62</a>>.<br>
Call-ID: <a href="mailto:8c977d261329cd80fd9d03d6@62.65.128.61">8c977d261329cd80fd9d03d6@62.65.128.61</a>.<br>
CSeq: 2 INVITE.<br>
User-agent: Netstream VoIP Gateway.<br>
Contact: <<a href="http://sip:0049xxxxxxxxxxx@62.65.128.62:5060" target="_blank">sip:0049xxxxxxxxxxx@62.65.128.62:5060</a>>.<br>
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,SUBSCRIBE.<br>
Content-Type: application/sdp.<br>
Content-Length: 584.<br>
Proxy-Authorization: Digest username="anonymous", realm="62.65.128.62",<br>
nonce="a4151ee0-1fbb-11de-b056-494b9de21e06", nc="00000001",<br>
uri="<a href="http://sip:0715aaaaaa@62.65.128.62:5060" target="_blank">sip:0715aaaaaa@62.65.128.62:5060</a>", cnonce="5f109eee",<br>
response="62faa6d38b3b12c3626753395a8b507c", algorithm="MD5", qop="auth".<br>
.<br>
v=0.<br>
o=- 225947743692042 1 IN IP4 62.65.128.62.<br>
s=-.<br>
c=IN IP4 62.65.128.62.<br>
t=0 0.<br>
m=audio 28224 RTP/AVP 8 18 4 3 100 100 99 100 100 98 97 96 105 0 101.<br>
a=rtpmap:8 PCMA/8000.<br>
a=rtpmap:18 G729/8000.<br>
a=fmtp:18 annexb=no.<br>
a=rtpmap:4 G723/8000.<br>
a=fmtp:4 annexa=no.<br>
a=rtpmap:3 GSM/8000.<br>
a=rtpmap:100 speex/8000.<br>
a=rtpmap:100 speex/8000.<br>
a=rtpmap:99 G726-16/8000.<br>
a=rtpmap:100 speex/8000.<br>
a=rtpmap:100 speex/8000.<br>
a=rtpmap:98 G726-24/8000.<br>
a=rtpmap:97 G726-32/8000.<br>
a=rtpmap:96 G726-40/8000.<br>
a=rtpmap:105 iLBC/8000.<br>
a=rtpmap:0 PCMU/8000.<br>
a=rtpmap:101 telephone-event/8000.<br>
a=fmtp:101 0-16.<br>
<br>
Best regards<br>
Peter<br>
<br>
<br>
<br>
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</blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
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