Thanks for the firewall hint. Actually, most of my UDP and TCP port are open.
I believe the problem is related to NAT. Here is the reasoning:
- I call from SJPhone using a real IP address (e.g. 200.22.33.44)
- The call reaches my router (with a real IP address of 200.33.44.55), which
routes the packets to my * server at the fake IP 192.168.0.200
- My * server places a call through my TDM interface
- I can hear on SJPhone the other party's voice, but he cannot hear me.
Taking a look at SIP debug info, I've found out that * get my IP address
correctly, but it sends to my SIP client the IP 192.168.0.200 address !!! So,
I guess my voice packets are going nowhere.
Any other hint?
Best,
Paulo Mannheimer
> -----Original Message-----
> From: asterisk-users-admin@lists.digium.com
> [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Andy Powell
> Sent: May 08, 2003 3:20 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] SIP-firewall problem?
>
> Paulo,
>
> You are going to have to open some ports for the RTP streams to get
> through.
> IIRC something like ports 16384 to 16391 which will also need to be
> forwared
> to your PC. I have had reports that just using 8000 and 8001 but I've
> been
> unable to replicate success with these 2 ports.
>
> HTH
>
> Andy
>
> *********** REPLY SEPARATOR ***********
>
> On 08/05/2003 at 02:35 Paulo H. Mannheimer wrote:
>
> >Hi,
> >
> >I have asterisk working well on my intranet, using SJPhone to make SIP
> >calls.
> >
> >Everything works fine except when I try to connect from outside my
> >intranet.
> >I've opened the 5060 UDP port on my D-link firewall/router and,
> although I
> >can
> >dial, complete calls and hear the other party, my voice gets lost (the
> >other
> >party does not hear me).
> >
> >I've took a look on the >sip debug output and everything seems
fine.
> Any
> >other
> >hint on what am I missing to configure on my firewall?
> >
> >Best regards,
> >
> >Paulo Mannheimer
> >
> >
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
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