[Asterisk-Users] Audio problems 50% of the time.

[Damon Estep]

A few things;

You have nat and qualify = yes, those settings are correct.

On your DSL, is there a public IP address on the internet side of the
Linksys? (not in the 10.x.x.x, 192.168.x.x, or 172.16.x.x subnets).

If not, you have another NAT router in the middle (your DSL modem) and
you will not have good luck.

The ATA186 is an antique in the voip world, it is no longer supported
and has poor and primitive SIP firmware. Spend $80-90 and get a Sipura
SPA2100. It is a combo router/ATA that works very well and has very
refined SIP images. Use 3.2.5(d) with asterisk 1.2 for best results.
Make sure you set the RTP packet size to .020 if you do use a SPA2100,
the default is .030 (30ms).

Make sure your Linksys router has current firmware. The newer (hardware
version 4) 4 and 8 port wired routers support QoS, as well as the
WRT54G. if you have one of those you should turn on the QoS, my guess is
you do not or the issue you report would likely not exist.

Try putting the ATA in the "DMZ" on the Linksys by setting the DMZ host,
but make sure your ata is secured with strong passwords since it will
become accessible from the internet.

The issue you are having is most likely NAT/Firewall related. When you
embrace newer technologies like VoIP you must also be willing to embrace
the cost of modern hardware, and the ATA186 does not fall into that
category... it was the device the Cisco/Linksys/Sipura folks cut thier
teeth on, and like many other 1st generation products, it sucks. The
same engineers that designed that device are still designing
Cisco/Sipura/Linksys devices; they just have a lot more experience now.





-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of kurt x
Sent: Wednesday, May 17, 2006 12:37 PM
To: Asterisk
Subject: [Asterisk-Users] Audio problems 50% of the time.

I have an Asterisk server that I use at work.  I have a phone that is
at home that logs into
the Asterisk server at work.  My home phone is hooked up via DSL
through a Linksys router. You can see the my sip.conf for the phone
blow.

The problem is each time the phone rings I can hear/be heard 50% of the
time.

Any suggestion on what to look for.

I do have my reg time set for 180 seconds on the cisco ATA186.

[72459]
type=friend
username=XXXXXX
secret=XXXXX
host=dynamic
context=voice-mail
dtmfmode=rfc2833
;canreivet=yes
nat=yes
qualify=yes

Kurt
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