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

[Joseph]

I can not help you much, but only confirm that SIP call from one of my provider in Poland is not working.
Registration goes through OK but call does not go through.
Back to 1.4 version is the solution.

--
Joseph

On 11/15/09 19:05, Eric van der Vlist wrote:
>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

-- 
Joseph