Hardphone Sip nat to nat

Joined
Mar 21, 2008
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Hi,
I install my elastix server at my work. I forward 5060 and 10000-20000 port to my elastix. I configure my sip_nat.conf nat=yes externip and localnet. All my local phone work fine. Now i install an 57i ct at home and i try to make call. All thing work but when the other people answer i have no sound. How can i debug this problem. Thank You!
 
Joined
Feb 17, 2007
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Hi I have the very same problem and it is now starting to get on my nerves!

Actually it is funny, because I can make a call from inside my lan to the elastix server to my voip provider (musimi.dk) to the pstn network and it is working perfectly - BUT if I make a call to another voip customer through musimi.dk and it is the very same phone he has connected to the PSTN network as well (siemens gigaset 550). Also it is impossible to maintain a phone registered for more than a few hrs. Even though I have set the phone to reregister every 60 seconds. Please if anyone have a smart way to get arround NAT problems please don't hessitate to write the wisdom. It is the only reason that I still have 3 pstn lines up and running.:blush:

Is seems to me that the NATTING "hides" the port information to asterisk and therefore the stream does not connect to the right line.

Rgds Michael
 
Joined
Mar 18, 2008
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hi bmartinp... maybe it's an udp problem... I use grandstream and the rtp port is setup at
port 5004 by defaut, so in my router I forwarded from 5004 to 5084 udp ports, and not only
port 5060... the same for 10000 to 20000 in the rtp.conf file I
started from 10001 to 20000 instead... I don't know if this helps you...
something you can do is a rtp debug in CLI so you can see what are the
udp ports that are used in the calls...

hope it helps...
 
Joined
Apr 15, 2008
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hi there,

I had the same problem at first but I found at that it was a router issue. I replaced the router(netgear) with linksys and installed the DD-RWT on it. It worked flawless after that. Also try to start rtp from 8000, instead of 10000. Change this in your rtp.conf file. Good luck,
 

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