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[asterisk-users] Behind NAT: source is fring software (SIP)


 
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bilmar_gh at yahoo.com
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PostPosted: Fri Jun 13, 2008 5:16 am    Post subject: [asterisk-users] Behind NAT: source is fring software (SIP) Reply with quote

Hi All;

My Asterisk is behind NAT with IP Address 192.168.0.2. I configued on my iPlanet router and port forwarding for 5060 (UDP) to be forwarded for 192.168.0.2 and I was able to let the fring softphone (SIP) to register on the asterisk.

But when caller initiate call, the caller hear the destination but the destination does not hear the caller.

I checked the RTP port range and I found it (10000 - 20000) and I forwarded it for the internal IP address 192.168.0.2 but the problem stayed!!

I do not know what should I do more? What it could be the reason for the problem? What should I do on the router more?

I am also thinking if the fring software could use UDP ports other than the range setted in the rtp.conf? Is it possible that source to use different port than the Asterisk RTP ports?

Note: do I have to do port forwarding on my router for the RTP UDP ports, or it is enough to forward the 5060 UDP port for the internal IP address 192.168.0.2?

Any help?

Regards
Bilal
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spamsucks2005 at gmail...
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PostPosted: Fri Jun 13, 2008 6:45 am    Post subject: [asterisk-users] Behind NAT: source is fring software (SIP) Reply with quote

Bilal,

where are you? I bought an Allnet from an importer here in France.
Otherwise, Germany or the UK will have them. Somewhere around 100 eu.

randy
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bilmar_gh at yahoo.com
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PostPosted: Fri Jun 13, 2008 6:46 am    Post subject: [asterisk-users] Behind NAT: source is fring software (SIP) Reply with quote

Dear Randy;

I am in Kuwait.
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