Sunday 13 July 2014

Panasonic ncp500 asterisk sip trunk no audio issue

Just a simple note to myself for the next time i forget, the dsp card, must be given an ip on the same network as panasonic.
In my scenario the ncp500 has a network address of 192.168.17.101
the dsp card should be 192.168.17.102, this will enable the codecs on the same network.

The setting then if a sip extension is setup on the ncp500 for example 404 on the asterisk piaf is as follows

incoming
username=404
type=peer
secret=404
qualify=yes
port=5060
nat=no
insecure=very
host=192.168.17.101
fromuser=403
dtmfmode=rfc2833
dtmf=rfc2833
allow=ulaw
disallow=all
context=from-trunk
canreinvite=no
maxexpiry=3600

outgoing

type=user
secret=404
allow=ulaw
disallow=all
context=from-trunk

then on inbound route for that trunk set
Pause Before Answer to the amount of time to ring before voicemail cuts in eg 17 seconds, about 3-4 rings