[asterisk-dev] Call-ID register

[Ismael Bouya]

(Note: in the time waiting to be registered to the list I found a
workaround, that is to set fromdomain to the correct domain. However I
think it remains a bug anyway that maybe could be corrected, certainly
from the registrar, but asterisk could do something about it too maybe?)

Hi,
I was running asterisk for more than 8 month without problem, and it
stopped working last saturday concerning the register command (I did
things on my server, but nothing concerning asterisk and the last upgrade
in my distribution was in late september and went well)

After a few tests and tcpdumps, I found that asterisk sent register
command, got an answer but didn't listen to it. Or it did listen to it (I
see it in the debug mode of the console), but doesn't treat it as an
answer: (ip and logins have been changed)
--- (9 headers 0 lines) ---
Retransmitting #1 (no NAT) to 198.151.12.54:5060:
REGISTER sip:sip.registrar.fr SIP/2.0
Via: SIP/2.0/UDP 198.51.100.24:5060;branch=z9hG4bK36a7dfbc
Max-Forwards: 70
From: <sip:0033999999999 at sip.registrar.fr>;tag=as09ca9745
To: <sip:0033999999999 at sip.registrar.fr>
Call-ID: 5d579b0674f45ba8347883985d553b64@[2001:db8:200:13:1::1]
CSeq: 103 REGISTER
User-Agent: Asterisk PBX 11.5.1
Expires: 120
Contact: <sip:Login at 198.51.100.24:5060>
Content-Length: 0

---

<--- SIP read from UDP:198.151.12.54:5060 --->
SIP/2.0 401 Unauthorized
Call-ID: 5d579b0674f45ba8347883985d553b64@[2001:DB8:200:13:1:0:0:1]
CSeq: 103 REGISTER
From: <sip:0033999999999 at sip.registrar.fr>;tag=as09ca9745
To: <sip:0033999999999 at sip.registrar.fr>;tag=00-08168-00d2af85-2af415a65
Via: SIP/2.0/UDP
198.51.100.24:5060;received=198.51.100.24;rport=5060;branch=z9hG4bK36a7dfbc
WWW-Authenticate: Digest realm="sip.registrar.fr",nonce="00d2ad3e462b32fb3b17a78e1258231a",opaque="00c677ab4f21d6a",stale=false,algorithm=MD5
Server: Cirpack/v4.42a (gw_sip)
Content-Length: 0


And asterisk keeps sending his register command, and never listen to the
answer.

The only explanation is the slight difference in the Call-ID's host name
in the answer (caps and 0's).
Is that important for asterisk? (that would correspond to the fact that
it appeared precisely last saturday since I worked on IPv6 at this time)

In that case, what can I do?

My "asterisk" realm is sip.mydomain.com, which is a cname to
mydomain.com, which has an AAAA entry since last weekend.

Is there a possibility to ask to asterisk not to change sip.mydomain.com
to the ip address in his call-ID? How? (I couldn't find that anywhere in
the documentation)

Of course I have no control on the registrar, so I cannot ask them to be
more conservative of the call-id

It looks like a rather new "bug", because I have a debian with exactly
the same configuration that works perfectly (it sends @domain2.com
instead of @ip.address), but with an older version of asterisk
(1.8.13.1~dfsg-3+deb7u1 on the old, 11.5.1 on the new)


Thanks in advance for your help!

Best regards,
-- 
Ismael
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