[Asterisk-Users] Phones can talk to asterisk but not each other through it

[WipeOut]

Tony Mountifield wrote:

>I posted this a week or two ago but no replies, so trying again...
>
>Summary: Two phones in different locations, each behind NAT, can both
>talk to an Asterisk server on the net, for the demo or for voicemail,
>but can't maintain a call to each other via that asterisk.
>
>Original post with details:
>
>I have a problem with an installation of asterisk on my colo server.
>I have a Grandstream BT102 behind a Linux NAT firewall, and my colleague
>also has one behind his.
>
>My connection is ADSL with 512k down and 256k up. My colleague's is
>Cable with 600k down and I don't know whether it's 128k or 256k up.
>
>I have the phones set up in sip.conf with nat=yes, qualify=yes and
>canreinvite=no. Each phone can successfully connect with Asterisk
>and dial the Asterisk Demo, leave and pick up voicemail, etc.
>
>However, if one phone tries to dial the other, once the called phone
>is answered, the audio starts off very stuttery and broken, and after
>a few seconds dies completely and the call gets dropped.
>
>In the asterisk log there are many entries for that time saying:
>Recv error: Resource temporarily unavailable.
>
>I am using the zaprtc timer module on the asterisk server, but in any
>case I understood that was only required for MeetMe or MOH.
>
>The server system is a Duron XP 1800, with 512MB RAM, running Fedora
>Core 1 with updates, and a standard 2.4.22 kernel that was recompiled
>only to make the RTC a module instead of compiled in (so I could rmmod
>it and then load zaprtc instead, which works fine).
>
>Can anyone suggest what things I should check or change?
>
>Cheers
>Tony
>  
>
Have you setup any port forwarding on the NAT boxes?
If not try it, it may help..