connection but no sound at remote extension

berend

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#1
Hi

When connecting to my elastix server with a remote sip phone I can make connections but no sound is transmitted.. I have configured NAT=yes for the extension and the externalip for the elastix server. All relevant ports are forwarded to the elastix server. When I move my remote extension to my office and don't change any settings (connection to the external ip address of the elastix server from inside) everyting works fine. I followed the suggestions on elastix without tears and as good as everything I could find on freepbx, asterisk and elastix form.

Can someone please help.

Cheers, Berend
 

dicko

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#2
One way sound or no-way sound?

Either way, it is almost certainly misconfigured PNATing either on the Elastix Box or the Router/firewall at either end of the connection if you have eliminated the Elastix box as a problem (localnet, externip/host etc.) then look at the routers between the server and the extension, there are many types of NAT'ing and some routers try to rewrite the rtp ports on the way through, if they do that then you are guaranteed no sound, so read carefully the documentation of the routers involved (both ends)

rtp debug will show the audio load and it should be bidirectional and symmetrical after a session is initiated.,
 

ramoncio

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#3
Usually this is a problem with your internet router. Which ports did you exactly open?
Also some internet providers block some ports, make sure yours doesn't.

Try to use 'rtp debug' as dicko wisely says, this will tell you if the audio gets to the Elastix box.
You can also use wireshark or tcpdump to capture all traffic and analize it. If you dump the traffic from a machine between your external router and your Elastix server your should see if the rtp passes through the router's nat for sure.
 

berend

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#4
I have used the rtp debug command at the asterisk cli. But, :blush: where do I find the rtp specific messages. All I get is the usual stuff that us in my asterisk log also but no rtp messages. Is there a specific rtp log?
 

berend

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#5

dicko

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#6
rtp debug

should show on the CLI if your verbosity is higher than 2 (and in /var/log/asterisk/full)

it looks like
Sent RTP packet to 192.168.1.221:3000 (type 00, seq 030476, ts 225600, len 000160)
Got RTP packet from n.n.252.23:28168 (type 00, seq 006054, ts 225760, len 000160)
Sent RTP packet to 192.168.1.221:3000 (type 00, seq 030477, ts 225760, len 000160)
Got RTP packet from n.n.252.23:28168 (type 00, seq 006055, ts 225920, len 000160)
Sent RTP packet to 192.168.1.221:3000 (type 00, seq 030478, ts 225920, len 000160)
Got RTP packet from n.n.252.23:28168 (type 00, seq 006056, ts 226080, len 000160)
Sent RTP packet to 192.168.1.221:3000 (type 00, seq 030479, ts 226080, len 000160)


it only will show if the session is opened, no rtp if no session.

use sip debug ip <ip of far end> to check that the call is actually completing.
 

berend

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#7
Hi dicko,

I have issued rtp debug at the asterisk cli. When making a call no RTP messsages appear. Also in the full log no RTP messages. Verbosity is set to 5.

Cheers, Berend
 

ramoncio

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#8
Hi berend,
You still haven't said which ports did you exactly open in the router... maybe you did something wrong.
If you only have Internet router - Elastix and you havent configured any firewall into the Elastix box, then the problem is with your router configuration.
It could also help if you said what router brand and model it is.

You can also try to set the DMZ host to the Elastix ip, if you router has that option, and then configure iptables at your Elastix box, so you only leave the needed ports open.
 

berend

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#9
Hi ramoncio,

I have forwarded SIP 5060 UDP and RTP 10000 - 20000 UDP and TCP
The router brand is Linksys but router OS is DDWRT.

Cheers, Berend
 

danardf

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#10
You don't need to open any rtp port with the TCP protocol.
Only use the UDP. ;)
 

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