[asterisk-users] Sip incoming call issue with Asterisk 1.6

[Eric van der Vlist]

After a migration to asterisk 1.6, I don't receive sip incoming calls
anymore.

As fas as I understand the SIP debug traces, my server receives the
request and reject it:

++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
<--- SIP read from UDP:212.27.52.5:5060 --->
INVITE sip:s at 192.168.4.2:5060;transport=udp SIP/2.0
Call-ID: 25151-WW-0eaf098b-2f615ac60 at freephonie.net
Contact: <sip:172.17.20.241:5062>
Content-Type: application/sdp
CSeq: 239836027 INVITE
From: "096160XXXX" <sip:096160XXXX at freephonie.net;user=phone>;tag=25151-GA-0eaf098c-32a97dc05
Max-Forwards: 28
Record-Route: <sip:C=on-88.165.134.117.5060;t=RDKIW at 212.27.52.5:5060;lr>
To: <sip:095199YYYY at 172.17.20.241;user=phone>
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-RDKI-00720204-55801b4f
Allow: UPDATE,REFER,INFO
User-Agent: Cirpack/v4.41c (gw_sip)
Content-Length: 173

v=0
o=cp10 125830752022 125830752022 IN IP4 212.27.52.129
s=SIP Call
c=IN IP4 212.27.52.129
t=0 0
m=audio 36480 RTP/AVP 8
b=AS:64
a=rtpmap:8 PCMA/8000/1
a=ptime:30

<------------->
--- (13 headers 9 lines) ---
  == Using SIP RTP CoS mark 5
Sending to 212.27.52.5 : 5060 (no NAT)
Using INVITE request as basis request - 25151-WW-0eaf098b-2f615ac60 at freephonie.net
Found peer 'freephonie_appelsortant' for '096160XXXX' from 212.27.52.5:5060
asterisk*CLI> 
<--- Reliably Transmitting (no NAT) to 212.27.52.5:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-RDKI-00720204-55801b4f;received=212.27.52.5
From: "096160XXXX" <sip:096160XXXX at freephonie.net;user=phone>;tag=25151-GA-0eaf098c-32a97dc05
To: <sip:095199YYYY at 172.17.20.241;user=phone>;tag=as03dcbe68
Call-ID: 25151-WW-0eaf098b-2f615ac60 at freephonie.net
CSeq: 239836027 INVITE
Server: Asterisk PBX 1.6.2.0~rc2-0ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="78360854"
Content-Length: 0
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ 

Some googling kind of suggest that this might be because for my ISP my
username is also my phone number:
http://lists.digium.com/pipermail/asterisk-dev/2009-January/036259.html

> The problem arises since you use phone numbers as identifiers for the  
> users. This is not a good thing (TM) and should be avoided. The  
> dialplan is where you route phone numbers. Devices should have device  
> names that you address in the dialplan on the extension that is  
> supposed to connect to one or several devices.

Am I right or must I search elsewhere?

Whether it's a good thing or not, I doubt I can convince Free
(http://free.fr) which is one of the biggest ISPs in France to change
their policy so that I can receive SIP calls again...

If my diagnostic is right, is there a way to work around this issue with
asterisk 1.6?

Thanks,

Eric

-- 
Eric van der Vlist <vdv at dyomedea.com>
Dyomedea (http://dyomedea.com)