[Kamailio-Users] openser and asterisk auth problem.

BERGANZ François francois at acropolistelecom.net
Thu Dec 18 10:31:18 CET 2008


-----I have that error:



<--- SIP read from UDP://192.168.1.156:5060 --->
INVITE sip:33662971130 at 192.168.1.156 SIP/2.0
Record-Route: <sip:192.168.1.156;lr=on;ftag=0c46027a>
Via: SIP/2.0/UDP 192.168.1.156;branch=z9hG4bK20d2.de427962.0
Via: SIP/2.0/UDP 192.168.1.60:43490;branch=z9hG4bK-d8754z-c15d2350d0742a11-1---d8754z-;rport=43490
Max-Forwards: 70
Contact: <sip:6833211245 at 192.168.1.60:43490>
To: "33662971130"<sip:33662971130 at 192.168.1.156>
From: "6833211245"<sip:6833211245 at 192.168.1.156>;tag=0c46027a
Call-ID: YTkxOWU5MGJjZjJhMWFmODZhMzZkMTU1YzUzYjUyMTc.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1100l stamp 47546
Content-Length: 231

v=0
o=- 1 2 IN IP4 192.168.1.60
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.1.60
t=0 0
m=audio 1402 RTP/AVP 0 101
a=alt:1 1 : Aov5DnDD ST5neNKz 192.168.1.60 1402
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (14 headers 10 lines) ---
Sending to 192.168.1.156 : 5060 (no NAT)
Using INVITE request as basis request - YTkxOWU5MGJjZjJhMWFmODZhMzZkMTU1YzUzYjUyMTc.
Found user '6833211245' for '6833211245'

<--- Reliably Transmitting (NAT) to 192.168.1.156:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.156;branch=z9hG4bK20d2.de427962.0;received=192.168.1.156
Via: SIP/2.0/UDP 192.168.1.60:43490;branch=z9hG4bK-d8754z-c15d2350d0742a11-1---d8754z-;rport=43490
From: "6833211245"<sip:6833211245 at 192.168.1.156>;tag=0c46027a
To: "33662971130"<sip:33662971130 at 192.168.1.156>;tag=as44aea1c2
Call-ID: YTkxOWU5MGJjZjJhMWFmODZhMzZkMTU1YzUzYjUyMTc.
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="653a3a8a"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'YTkxOWU5MGJjZjJhMWFmODZhMzZkMTU1YzUzYjUyMTc.' in 32000 ms (Method: INVITE)
asteriskfcois*CLI>
<--- SIP read from UDP://192.168.1.156:5060 --->
ACK sip:33662971130 at 192.168.1.156 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.156;branch=z9hG4bK20d2.de427962.0
From: "6833211245"<sip:6833211245 at 192.168.1.156>;tag=0c46027a
Call-ID: YTkxOWU5MGJjZjJhMWFmODZhMzZkMTU1YzUzYjUyMTc.
To: "33662971130"<sip:33662971130 at 192.168.1.156>;tag=as44aea1c2
CSeq: 2 ACK
Max-Forwards: 70
User-Agent: Kamailio (1.4.1-notls (i386/linux))
Content-Length: 0





-------Sip.conf:

[6833211245]
type=friend
username=6833211245
accountcode=6833211245
regexten=6833211245
callerid=6833211245
amaflags=billing
secret=2598992992
nat=yes
insecure=invite
dtmfmode=RFC2833
qualify=yes
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g729
host=dynamic
context=a2billing
regseconds=0
cancallforward=yes

[ser_mnup]
type=peer
qualify=yes
disallow=all
allow=ulaw
allow=alaw
dtmfmode=rfc2833
canreinvite=no
insecure=invite
progressinband=yes
context=mnup
host=192.168.1.156






Cordialement,
BERGANZ François


http://www.acropolistelecom.net
 Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

-----Message d'origine-----
De : users-bounces at lists.kamailio.org [mailto:users-bounces at lists.kamailio.org] De la part de Andrew V.Statsenko
Envoyé : jeudi 18 décembre 2008 10:15
À : users at lists.kamailio.org
Objet : ***SPAM*** Re: [Kamailio-Users] openser and asterisk auth problem.

В Чтв, 18/12/2008 в 09:41 +0100, BERGANZ François пишет:
> Hello,
 
> 
> I have :
> 
> ASTERISK-----OPENSER-----SOFTPHONE
> 

> My softphone auth to openser (with username/password of asterisk sync
> with a database…)
> Openser forward the INVITE to asterisk and asterisk return Unautorize!
> I tried to have the same realm to try de have the same auth for
> asterisk and openser…
> 
> But without success.

> In fact, I need to auth to openser(with asterisk config user in the
> database) and forward to asterisk the INVITE.


First off all try to read sip.conf &
http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf 

Second.. you have to configure OpenSER as _peer_ in Asterisk
configuration ;-) 


[openser]
type=peer
host=IP_OR_FQDN_HERE
....


--
WBR,
Alter




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