No audio on SIP calls with Asterisk

The connection is Asymmetric. I have set QoS for sip and RTP to premium. I don't think it's happening only when the outgoing traffic is saturated.

Setup
PBX-in a flash

Trunk 1 setup
qualify=3600
nat=yes
insecure=very
host=chicago-1f.vtnoc.net
fromdomain=chicago-1f.vtnoc.net
context=from-trunk&from-trunk
canreinvite=no
authuser=xxxxxxxxxx
dtmfmode=inband

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Trunk 2 setup
type=peer
username=xxxxxx
disallow=all
allow=ulaw
; allow=g729
fromuser=149830
trustrpid=yes
sendrpid=yes
insecure=invite
qualify=yes

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Attached is the log for the call. It is abnormally large. For a 5 second call the log was almost 7000 lines!

-Jaga
asterisk-log.txt
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Jeff swicegoodTechnicianAsked:
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José MéndezConnect With a Mentor Commented:
Got it. Asterisk is the only one sending audio to IP address 216.246.105.146. I was also able to listen to Jitsi's audio stream, heard your prayers by the way, nice singing mate.

This means the audio stream from the phone to Asterisk, and from Asterisk to the provider is fine in terms that it exists at the minimum. The other way around doesn't, however.

There are no audio packets from the provider's IP address. They advertised that IP

Connection Information (c): IN IP4 216.246.105.146

would handle the audio, yet there is no trace of it in the capture, from Asterisk's standpoint.

When Asterisk sent them the connection info, it went out as follows:

Connection Information (c): IN IP4 76.6.24.151

So this is Asterisk telling the provider where to send the audio on the public network. Now the questions are:

- Is that correct?
- If so, what device in your network is in charge of NAT'ing IP  76.6.24.151?
- Is it logging RTP activity from the provider?

1 more clarification: is this dead air between you and the far end party, or is it 1 way audio?
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José MéndezCommented:
tcpdump -w /tmp/1wayaudio.cap  --> reproduce the affected call while running, then finish with ctrl-c

then upload to cloudshark.org so we can analyze the situation.

Thanks,
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Jeff swicegoodTechnicianAuthor Commented:
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José MéndezCommented:
Apologies, think the -s switch is needed for packets not to be limited in size:

tcpdump -w /tmp/1wayaudio.cap -s 0

Also calling and called numbers will be needed to identify the call flow.

Thank you,
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Jeff swicegoodTechnicianAuthor Commented:
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Jeff swicegoodTechnicianAuthor Commented:
Ok, problem solved. The device that NAT's the external IP has only one client anther router with another NAT. I put that client in the DMZ, eliminating the double NAT and it worked. Thanks a lot. Hare Krishna!
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José MéndezCommented:
Glad to hear that!
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