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PBX stops sending audio mid-sentence (of outgoing message)

Posted on 2013-11-22
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Asterisk: PBX stops sending audio mid-sentence (of outgoing message). I don't think it's happening because the outgoing traffic is saturated.

I have attached the logfile. In there is says that  the transmission is not being responded to. This is not a problem of 1 way audio. When it works, it works both ways.

Sometimes it does not stop. Maybe this is an issue for Viatalk, my provider.
asterisk-stops-sending-audio-mid.txt
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Question by:Jeff swicegood
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by:Jeff swicegood
ID: 39678759
The PBX is behind a router which is behind a router/modem. I have put the router in the DMZ of the modem to eliminate the double NAT. Outgoing calls never drop connection with the peer (provider).

username=1919XXXXXXX
type=peer
secret=XXXXXXXXXX
qualify=3600
nat=yes
insecure=very
host=chicago-1f.vtnoc.net
fromdomain=chicago-1f.vtnoc.net
context=from-trunk&from-trunk
canreinvite=no
authuser=1919XXXXXXX
dtmfmode=inband

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Could it be that the PBX is not reading the correct external IP?

Here is a picture of my SIP configuration.SIP settingsSIP settings
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Jeff swicegood earned 0 total points
ID: 39707717
Found the following online, applied it and it worked!

It's Your Firewall, Stupid. I wish I had a nickel for every message thread that has been written that goes something like this. "I can make calls out of my system, but the people I call can't hear me." Or vice versa. The answer is pretty simple if you stop and think about it for a second. A phone call has two participants. One talks and the other one listens. Then you take turns. At least that's the theory. For that to actually work in the world of Internet telephony, the talking legs of the call have to be able to get from Point A to Point B and from Point B to Point A. If your IP-based telephone or Asterisk system is sitting behind a firewall/router, you have to configure your router to pass the incoming data into the server and telephone on your private network. If the telephone or Asterisk system on the other end of the call happens to also be sitting behind a firewall/router, then we have what's called "double NAT issues." And, no, this doesn't refer to no-see-ums on a steamy summer night in Dixie. Bottom line: If any of this communications traffic can't find it's way to the other end, then someone can't hear all or part of the telephone conversation.

To fix NAT problems with Asterisk, you simply tell your router to forward all data received on UDP ports 4569, 5004 to 5037, 5039 to 5082, and 10000 to 20000 to the private IP address of your Asterisk server. You also must make certain that the following entries exist in /etc/asterisk/rtp.conf:

[general]
rtpstart=10000
rtpend=20000

And bindport = 5060 must exist in the [general] context of /etc/asterisk/sip.conf. The aggregations take care of the rtp.conf and sip.conf setups for you. But you must reconfigure your router/firewall. Last, but not least, you probably need to complete the next step below as well.
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