Remote Extension incoming/outgoing no voice heard

japok25

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#1
The asterisk server w/ a router box at the office works perfectly suddenly the hardware crashes I have backup setting of elastix so I tried to restore it into another server with the same version of elastix 1.6.2-1 and I think it restored all the settings. Now I buy a new router and the brand is TRENDnet TW100-BRF114 I have already open all the ports of the router. Then I test it locally using ATA’s and Softphones where SIP Extension incoming and outgoing calls the voice it is works fine and now I test it remotely using Softphones same procedure incoming and outgoing calls it can ring anyone but you cannot hear any voice both ways.

Also I already tried to change the Sip_Nat.conf
nat=yes
externip=xxx.xxx.xxx.xx (external static ip at office)
localnet=172.16.2.0/255.255.255.0 (local network at office)


Can anyone have any clues that may fix the problem?
 

circuitid

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Dec 23, 2010
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#2
I would recommend looking at your firewall settings. Note that the SIP protocol uses port 5060 for signaling and RTP ports 8766 - 35000. If you can hear audio internally but not externally then it most likely has to do with the RTP ports being blocked to the external network.

For NAT and VoIP, I would also recommend looking at this article:
http://www.voip-info.org/wiki/view/NAT+and+VOIP
 

dicko

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#3
A quick note for those setting up their NAT routers.

Although 5060 is the default and conventional port to use, some would suggest that it is a security risk, it can easily be changed in

/etc/asterisk/sip.conf (or one of it's included files)

with

bindport=50XX

Your external extensions and VSP must of course agree on this.

the open ports for rtp (the audio load) are set in

/etc/asterisk/rtp.conf

by default they are

rtpstart=10000
rtpend=20000

your firewall needs only match what you have here.
 

fmvillares

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#4
hi my great friend dick!!! 1 add...udp only by default...tcp is not used by default in asterisk (but could be used in sip since 1.6.x)
 

dicko

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#5
Fernando, my friend and fellow black sheep :) welcome back . . .

Of course no tcp needed for rtp ;)

regards

Dick
 

japok25

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#6
Gentlemen,

Thank you very much for giving an ideas on how to passthrough the NAT routers. I figure it out how to configure my router TRENDnet TW100-BRF114 In this kind of router the Port Forwarding is a virtual server so I created a SIP Server 5004 – 5082 and RTP Server 8766 – 35000. When I test it in a remote extension it works thank again to all of you.

Here the link for your reference: http://trendnet.com/kb/kbp_viewquestion ... &catId=193


Regards to all,

Japok
 

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