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Jeff swicegood

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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
Avatar of José Méndez
José Méndez

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,
Avatar of Jeff swicegood

ASKER

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,
ASKER CERTIFIED SOLUTION
Avatar of José Méndez
José Méndez

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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!
Glad to hear that!