asterisk users - May 2004 - * and sip proxy auth

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is it possible to use PROXY AUTH with */sip ?

Szenario:

UAC => Asterisk ( SIP REGISTRAR/PROXY) => SER ( PROXY ) => Gateway.

while trying to call everythings works fine until
* sends the invite to the ser proxy. This asks for a PROXY AUTH.
* then sends as user the "called Phonenumber" and the proxy denies.

i tried to call 7777923 which should be router to the gateway.

UAC => *

-> INVITE
-> trying 

* => SER

-> INVITE
-> 407 Proxy Auth
-> ACK
-> INVITE sip:7777923@domain.com SIP/2.0
Via: SIP/2.0/UDP 10.10.10.184:5060;branch=z9hG4bK4863727a
From: "8009090" <sip:8009090@10.10.10.184>;tag=as7eed983e
To: <sip:7777923@domain.com>
Contact: <sip:8009090@10.10.10.184>
Call-ID: 290c4802496784155d18c21e21dfd1ad@10.10.10.184
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username="7777923",
realm="10.10.10.184",
                            ^^^^^^^^^^^^^^^^^^
algorithm="MD5", uri="sip:7777923@domain.de",
nonce="40a252a3c2e6a78c7a9e4b135e1bc3df104225ed",
response="bb563767d727fbd7e89356886bfdf5fe", opaque=""
Date: Wed, 12 May 2004 16:30:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 316

v=0
o=root 13973 13975 IN IP4 10.10.10.184
s=session
c=IN IP4 10.10.10.184
t=0 0
m=audio 13828 RTP/AVP 8 0 18 3 10 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=rtpmap:10 L16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

-> SIP/2.0 407 Proxy Authentication Required
-> ACK

Proxy Server denied access.


sip.conf:

[general]
port = 5060
bindaddr = 0.0.0.0
context = incoming
disable=all
allow=alaw
allow=alaw
allow=ulaw
allow=g729
allow=gsm
allow=slinear
srvlookup=yes
canreinvite=yes

register => 8009090:passwd@domain.com/8009090

[100]
type=friend
context=out
callerid=<100>
username=Markus
host=dynamic
dtmfmode=info                   ;or inband if you prefer
secret=100
qualify=5000
mailbox=100
canreinvite=no                  ;As long as the phones are NATed


Any Ideas? Thanx a lot,

Markus

Sounds like you are using CVS 1.0stable?

Proxy Authentication is broken in that CVS head, and it may not get fixed.
Using development head will fix this.
see also the bugtracker.




is it possible to use PROXY AUTH with */sip ?

Szenario:

UAC => Asterisk ( SIP REGISTRAR/PROXY) => SER ( PROXY ) => Gateway.

while trying to call everythings works fine until
* sends the invite to the ser proxy. This asks for a PROXY AUTH.
* then sends as user the "called Phonenumber" and the proxy denies.

i tried to call 7777923 which should be router to the gateway.

UAC => *

yes i am using stable,

ok i will try unstable,

thanx a lot ,

Markus

Am Mit, den 12.05.2004 schrieb Karl Brose um 18:55:
> Sounds like you are using CVS 1.0stable?
> 
> Proxy Authentication is broken in that CVS head, and it may not get fixed.
> Using development head will fix this.
> see also the bugtracker.
> 
> 
> 
> 
> is it possible to use PROXY AUTH with */sip ?
> 
> Szenario:
> 
> UAC => Asterisk ( SIP REGISTRAR/PROXY) => SER ( PROXY ) =>
Gateway.
> 
> while trying to call everythings works fine until
> * sends the invite to the ser proxy. This asks for a PROXY AUTH.
> * then sends as user the "called Phonenumber" and the proxy
denies.
> 
> i tried to call 7777923 which should be router to the gateway.
> 
> UAC => *
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
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>    http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
[netzquadrat]GmbH    fon 0211.30 20 33 27
Ronsdorfer Str. 74   fax 0211.30 20 33 22
40233 D?sseldorf     mobil 0163 7777 923
weitere Info unter:  http://www.netzquadrat.de

** NEU von [NQ] - http://www.VOICEMEETING.de **
** die superg?nstige Sofort-Telefonkonferenz **