[Asterisk-Users] Asterisk with Sipbroker calling / routing problem

[Mathieu Chouquet-Stringer]

	Hello all,

I've been using * for quite some time and yesterday I decided to add
sipbroker to my config.  It was pretty simple and it works for some
numbers (e.g. I can call *258-9123, UK date & time - which is on the
"phone numbers you can call" page -) but fails for some others.

For example I've got a friend who's at freephonie so to call him, I
would dial *759608xxxxxxxx (7596 being freephonie.net).

When I do that, I get the following error:
Jun 29 10:27:21 NOTICE[7916]: chan_sip.c:9685 handle_response_invite: Failed to authenticate on INVITE to '<sip:0001 at somehost.somedomain.tdl>;tag=as32d2cdfe'

And here's a snippet of what I get from 'sip debug':

--------------------------------------------------------------------------

<-- SIP read from 24.196.79.163:5060: 
SIP/2.0 407 authentication required
Allow: UPDATE,REFER
Call-ID: 73833b4d1ddb389d7a8114e4684f091b at somehost.somedomain.tdl
Contact: <sip:212.27.52.5:5060>
CSeq: 102 INVITE
From: <sip:0001 at somehost.somedomain.tdl>;tag=as32d2cdfe
Proxy-Authenticate: Digest
realm="freephonie.net",nonce="012dd3995b84e8f56ca34a7201a0c6ff",opaque="012daad2220ed2c",stale=false,algorithm=MD5
Record-Route: <sip:24.196.79.163;lr;ftag=as32d2cdfe>
Server: Cirpack/v4.40 (gw_sip)
To: <sip:*759608xxxxxxxx at sipbroker.com>;tag=01-08146-012dd3ab-3b2383163
Via: SIP/2.0/UDP
172.16.1.1:5060;received=86.216.233.69;rport=5060;branch=z9hG4bK76bd560d
Content-Length: 0


--- (12 headers 0 lines)---
Transmitting (no NAT) to 24.196.79.163:5060:
ACK sip:*759608xxxxxxxx at sipbroker.com SIP/2.0
Via: SIP/2.0/UDP 172.16.1.1:5060;branch=z9hG4bK76bd560d;rport
From: <sip:0001 at somehost.somedomain.tdl>;tag=as32d2cdfe
To: <sip:*759608xxxxxxxx at sipbroker.com>;tag=01-08146-012dd3ab-3b2383163
Contact: <sip:0001 at 172.16.1.1>
Call-ID: 73833b4d1ddb389d7a8114e4684f091b at somehost.somedomain.tdl
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Jun 29 10:27:21 NOTICE[7916]: chan_sip.c:9685 handle_response_invite:
Failed to authenticate on INVITE to
'<sip:0001 at somehost.somedomain.tdl>;tag=as32d2cdfe'
Transmitting (NAT) to 172.16.1.19:5060:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP
172.16.1.19:5060;branch=z9hG4bK9954d222975cdcc1;received=172.16.1.19
From: <sip:0001 at asterisk;user=phone>;tag=2858979361
To: <sip:*759608xxxxxxxx at asterisk;user=phone>;tag=as4eecd6f3
Call-ID: 4074108287 at 172.16.1.19
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:*759608xxxxxxxx at 172.16.1.1>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


--------------------------------------------------------------------------


Here's what I got in sip.conf for sipbroker:
[sipbroker-out]
type=peer
fromuser=0001
fromdomain=somehost.somedomain.tdl
host=sipbroker.com
port=5060
canreinvite=yes
qualify=yes


Any idea what's going on?  I've been reading quite a few papers about
SIP authentication but I still fail to understand what's really
happening (or is freephonie not 'open')?

Any help is welcome!

Cheers,

-- 
Mathieu Chouquet-Stringer                         ml2news at free.fr