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Asterisk one way audio, outbound calls, only on phones sharing LAN with asterisk

Posted on 2009-06-29
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Last Modified: 2013-11-12
I am getting one way audio with a Linksys spa942, that is on the same LAN as asterisk.  The one way audio, seems to be only on outbound calls.  And the audio that is heard, is what is coming from the SPA942.  The inbound audio to the SPA942, is not heard.

II have about 15 phones across the WAN, that use the same asterisk server, but they have no audio issues.  I have tried pointing the phone to the internal private ip address of asterisk, as well as the external public ip address.

I have port 5061 forwarded to the phone, and the phone set to register and listen on 5061.  Asterisk is listening on 5060 and port 5060 is forwarded to asterisk.  I have udp & tcp ports 10000-20000 forwarded to asterisk.  

I am using a new Netgear FVS338 firewall, NAT enabled, with the second to latest firmware 3.0.4-19.  The most recent firmware 3.0.5-29 has issues with SIP.  This firewall replaced a Netgear FVS114, which is a small firewall, and one of their older models, which for me had zero problems, except it lacked throughput above 5mbps.  Like I said, the FVS114 had no audio problems.

I have sip_nat.conf and sip.conf, configured with
nat=yes
externip = 75.xxx.xx.xx
localnet = 192.168.0.0/255.255.255.0

Any ideas?
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Question by:jkockler
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6 Comments
 
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Accepted Solution

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Ron Malmstead earned 500 total points
ID: 24738265
Do you have "canreinvite=nonat" set ?

Look here...
http://www.voip-info.org/wiki/view/Asterisk+SIP+canreinvite
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Expert Comment

by:Ron Malmstead
ID: 24738275
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by:jkockler
ID: 24738343
I do not have the command listed at all.  What is the default, if not listed?
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Expert Comment

by:Ron Malmstead
ID: 24738518
canreinvite is an option in the SIP peer definitions..

It basically tells Asterisk if the phones are allowed to have direct RTP setup (phone to phone or phone to "provider"), rather than Asterisk staying in the media path.  I believe the default is yes, and in a NAT situation this could be problematic.

You are using a sip provider, correct ?
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by:jkockler
ID: 24738664
I set canreinvite=no and all issues resolved!!!!!!!!!!!!!!!  Thanks xuserx!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
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by:Ron Malmstead
ID: 24738712
Glad to help.
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