Hello Matthew,

My version is Asterisk 1.6.2.9.

Or have you seen NAT? I have no NAT on my network. Have you seen my little
diagram above?

Here it is:

SIP friends (phones) <-----> Asterisk <-----> SIP gateway to PSTN converter
80.236.215.61                109.69.217.6     internal IP (
10.4.0.10/255.255.255.0 )

My Asterisk server has two NIC/interfaces.

- 1 interface with public IP (109.69.217.6 to talk with SIP friends)
- 1 interface with internal ip (10.4.0.1 to talk with SIP gateway's)

SIP friend should not even know that the call is routed to the SIP/PSTN
gateway.
It could be a SIP trunk to a SIP provider Internet, the user does not have to
know...

Best regards,
Mickael



2013/6/13 Matthew J. Roth <mr...@imminc.com>

> Mickael MONSIEUR wrote:
> >
> > I have a standard Asterisk configuration:
> >
> > SIP friends (phones) <-----> Asterisk <-----> SIP gateway to PSTN
> converter
> > 80.236.215.61                109.69.217.6     internal IP (
> 10.4.0.10/255.255.255.0 )
> >
> > When analyzing traffic on a SIP friend/phone I see this:
> >
> > INVITE sip:xxxx@80.236.215.61:64946;ob SIP/2.0
> > Via: SIP/2.0/UDP 109.69.217.6:5060;branch=z9hG4bK52d50250;rport
> > Max-Forwards: 70
> > From: < sip:xxxx@109.69.217.6 >;tag=as15b47581
> > To: "test" < sip:xxxx@109.69.217.6>;tag=kp1VwHD80rA9MVdBjTF4jyFIaCkrJcjh
> > Contact: < sip:xxxxx@109.69.217.6 >
> > Call-ID: MSMhw2bsheHWAQgHlae3O7yKQ2P9EcsM
> > CSeq: 102 INVITE
> > User-Agent: Asterisk
> > Require: timer
> > Session-Expires: 1800;refresher=uas
> > Min-SE: 90
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> > Supported: replaces, timer
> > Content-Type: application/sdp
> > Content-Length: 217
> >
> > v=0
> > o=root 664087974 664087976 IN IP4 10.4.0.10
> > s=Asterisk
> > c=IN IP4 10.4.0.10
> > t=0 0
> > m=audio 8652 RTP/AVP 8 101
> > a=rtpmap:8 PCMA/8000
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-16
> > a=ptime:20
> > a=sendrecv
> >
> > My equipement IP 10.4.0.10 is visible to the user, why?
>
>
> Mickael,
>
> What version of Asterisk are you running?
>
> Is the Asterisk server outside and the SIP gateway to PSTN converter
> inside of a
> NAT?
>
> What are the NAT SUPPORT and MEDIA HANDLING settings in sip.conf?
>
> Regards,
>
> Matthew Roth
> InterMedia Marketing Solutions
> Software Engineer and Systems Developer
>
> --
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