Linksys SPA941 Phones...

mediumgrade
Posts: 15
Member Since:
2008-01-31

I have the Linksys SPA941 phones which, generally, work great.

However, when trying to get them to work remotely (the whole SIP/NAT thing), I get intermittent audio problems. Namely, neither party can hear the other. Now, I know that the NAT/Asterisk settings are correct as it works half the time with the Linksys phone and all the time with my X-Lite softphone, but I wondering if anyone knows anything specific to the SIP settings on the Linksys phone that I should additionally be aware of.

Here are my port forward settings:

22 <--- TCP
80 <--- TCP
443 <--- TCP
5060-5082 <--- TCP/UDP
8000-20000 <--- TCP/UDP

The first three, obviously, are SSH, HTTP and HTTPS, but the last two should be correct for SIP/RTP. The RTP ports for the Linksys phone are between 16384 and 16482 (well within what I have open on the firewall). What should I be looking into for this?

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David Martinez
IT Operations
Parker & Morgan
life is better



kerryg
Posts: 5533
Member Since:
2006-05-31
What do you have for your

What do you have for your sip_nat.conf settings?



mediumgrade
Posts: 15
Member Since:
2008-01-31
externip=65.255.194.171 local

externip=65.255.194.171
localnet=172.16.50.0/255.255.255.0

Now, I have all of the phones connecting to a host name which points to that IP address. This is done so that if/when we switch providers, it will be a smooth transition for all remote phones.

Is this not the way to do it?

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David Martinez
IT Operations
Parker & Morgan
life is better

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David Martinez
IT Operations
Parker & Morgan
life is better



eeknz
Posts: 62
Member Since:
2006-08-13
You need to set a stun

You need to set a stun server on the phone. Look at the bottom of the SIP page. Enable it and point it to a stun server. you could try stun.fwdnet.net if you can't find any others.



mediumgrade
Posts: 15
Member Since:
2008-01-31
My Apologies, I am new to

My Apologies, I am new to Asterisk/SIP technology. What exactly is STUN? If I set the Netgear to use this STUN server, is there anything I have to change on my Asterisk box?

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David Martinez
IT Operations
Parker & Morgan
life is better



olamsined
Posts: 2
Member Since:
2007-07-01
Alernative route

In your extension setting for those extensions.

Try setting Qualify to no instead of yes, it worked for me.

Qualify=no



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