[asterisk-users] SIP trunk between two Asterisk servers

[Vieri]

--- On Wed, 5/12/10, Philipp von Klitzing <klitzing at pool.informatik.rwth-aachen.de> wrote:

> > <--- SIP read from 192.168.250.111:5060 --->
> > SIP/2.0 407 Proxy Authentication Required
> 
> You need to run the SIP debug on 192.168.250.111 to learn
> more about WHY 
> the 407 is issued. Have a close look and you are likely to
> understand it 
> right away.
> 
> Also: Do not forget the "reload" after applying changes to
> sip.conf.

I always do a "sip reload" after changes to sip settings.

Here are the SIP messages on 192.168.250.111 (Asterisk server 1 - receiving end):

<-- SIP read from 192.168.250.112:5060:
INVITE sip:3666 at 192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport
From: "device" <sip:4053 at 192.168.250.112>;tag=as18a568d6
To: <sip:3666 at 192.168.250.111>
Contact: <sip:4053 at 192.168.250.112>
Call-ID: 328617546726e5d430538e80617716e1 at 192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 12 May 2010 09:20:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
upported: replaces
Content-Type: application/sdp
Content-Length: 270

v=0
o=root 20611 20611 IN IP4 192.168.250.112
s=session
c=IN IP4 192.168.250.112
t=0 0
m=audio 14648 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

--- (14 headers 13 lines) ---
Using INVITE request as basis request - 328617546726e5d430538e80617716e1 at 192.168.250.112
Sending to 192.168.250.112 : 5060 (NAT)
Reliably Transmitting (NAT) to 192.168.250.112:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;received=192.168.250.112;rport=5060
From: "device" <sip:4053 at 192.168.250.112>;tag=as18a568d6
To: <sip:3666 at 192.168.250.111>;tag=as57a19dac
Call-ID: 328617546726e5d430538e80617716e1 at 192.168.250.112
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1327c5b6"
Content-Length: 0


---
Scheduling destruction of call '328617546726e5d430538e80617716e1 at 192.168.250.112' in 15000 ms
Found user '4053'

<-- SIP read from 192.168.250.112:5060:
ACK sip:3666 at 192.168.250.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport
From: "device" <sip:4053 at 192.168.250.112>;tag=as18a568d6
To: <sip:3666 at 192.168.250.111>;tag=as57a19dac
Contact: <sip:4053 at 192.168.250.112>
Call-ID: 328617546726e5d430538e80617716e1 at 192.168.250.112
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

Can you deduce from this what I'm doing wrong?

Thanks,

Vieri