asterisk users - Nov 2010 - Ekiga can register but not my IP phone

Hello,

I have a Sip phone (Siemens C470IP) which works perfectly with
different VoIP providers (iptel, betamax, ovh...). It also worked well
with my testing server (ubuntu and inside the LAN).

But now the problem i have is that the hardphone doesn't connect to my
dedicated server (debian lenny / Asterisk 1.6.2.13). The strange thing
is that ekiga can connect to the same asterisk server with the same
SIP account.

Here is a part of my sip.conf :

    [general]
    dtmfmode=auto
    language=fr ; pour les messages lus par asterisk
    disallow=all
    allow=ulaw
    allow=alaw
    allow=speex

    [siemens]
    type=friend
    context=interne
    host=dynamic
    secret=xxxxxxxx

When i'm doing a sip set debug ip XXX.XXX.XXX.XXX i have some
information. It seems that asterisk receives the rengistration request
but doesn't answer to it. Here are the logs :
http://server.chab.info/Registration_logs_ip_phone.txt

Using Ekiga with the same SIP account (name is siemens) and from the
same physical location works well :
http://server.chab.info/Registration_logs_ekiga.txt

I didn't change anything about asterisk config (except sip.conf and
extensions.conf).
If you have any idea, please share it with me, i really don't to do to
fix this problem...
Thanks in advance !

On Fri, Nov 19, 2010 at 7:28 AM, Benoit Chabrier <chab at chab.info>
wrote:
> Hello,
>
> I have a Sip phone (Siemens C470IP) which works perfectly with
> different VoIP providers (iptel, betamax, ovh...). It also worked well
> with my testing server (ubuntu and inside the LAN).
>
I am assuming you mean Asterisk on Ubuntu inside the LAN
> But now the problem i have is that the hardphone doesn't connect to my
> dedicated server (debian lenny / Asterisk 1.6.2.13). The strange thing
> is that ekiga can connect to the same asterisk server with the same
> SIP account.
>
Is this outside the LAN?
Is there NAT in between?
SIP is a pain in the ass with NAT, so it's the only thing I can think
of. Usually in my experience it's the other way around! Ekiga is the
one that doesn't work and tends to be very quirky (takes a long time
to quit, has strange registration quirks, etc.), I mean when compared
to HW SIP device.
> Here is a part of my sip.conf :
>
> ? ?[general]
> ? ?dtmfmode=auto
> ? ?language=fr ; pour les messages lus par asterisk
> ? ?disallow=all
> ? ?allow=ulaw
> ? ?allow=alaw
> ? ?allow=speex
>
> ? ?[siemens]
> ? ?type=friend
> ? ?context=interne
> ? ?host=dynamic
> ? ?secret=xxxxxxxx
>
> When i'm doing a sip set debug ip XXX.XXX.XXX.XXX i have some
> information. It seems that asterisk receives the rengistration request
> but doesn't answer to it. Here are the logs :
> http://server.chab.info/Registration_logs_ip_phone.txt
>
> Using Ekiga with the same SIP account (name is siemens) and from the
> same physical location works well :
> http://server.chab.info/Registration_logs_ekiga.txt
>
> I didn't change anything about asterisk config (except sip.conf and
> extensions.conf).
> If you have any idea, please share it with me, i really don't to do to
> fix this problem...
> Thanks in advance !
The only thing I can think of are NAT issues with SIP. If you are in
fact NATing try the Siemens phone to a direct IP to the server (no
NAT, firewall, etc.) and see.

--
Alejandro Imass

Thanks for your help.

you were right it also work without a stun server adding to sip.conf:
externip=78.47.x.x  ; in [general] the IP of the dedicated server
nat=yes  ; in the description of my peer


2010/11/19, Alejandro Imass <ait at p2ee.org>:
> On Fri, Nov 19, 2010 at 8:23 AM, Benoit Chabrier <chab at chab.info>
wrote:
>> Thanks Alejandro, you were right it was just a NAT problem ! i add a
>> stun server in the phone configuration and it works :)
>>
>
> Cool. Also Asterisk SIP conf file has some NAT settings as well that
> you can play with and perhaps do away with the stun server config in
> the phone. Here is a great article that explains in detail the issues
> with SIP and NAT: http://www.voipuser.org/forum_topic_7295.html
>
>> 2010/11/19, Alejandro Imass <ait at p2ee.org>:
>>> On Fri, Nov 19, 2010 at 7:28 AM, Benoit Chabrier <chab at
chab.info> wrote:
>>>> Hello,
>>>>
>>>> I have a Sip phone (Siemens C470IP) which works perfectly with
>
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