[asterisk-users] Muted sound on a Linksys 962

[Danny Nicholas]

This worked for me
Exten => s,1,Answer()
Exten => s,n,Dial(Zap/g1/w5551212)

What happens is that * doesn't go "full duplex" until it does a "Native
Bridge".  The Answer Command creates a temporary bridge until the real one
can take effect.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of James Lamanna
Sent: Tuesday, January 27, 2009 12:38 PM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Muted sound on a Linksys 962

Hi,
One of our customers has an issue with the callee not being able to hear
them.
It seems to happen very frequently on one number in particular where
there are about 3 IVR menus to dial through
before getting to a live person. However, this does not happen on every
call.
Running tcpdump on the RTP packets, I can see that RTP is setting
sent, but the values in the packet
are all very close to 0xFF or 0x7F (which is 0 or -1 once you
translate it using G.711).
Could this be some issue with the phone muting audio because it's
"stuck" sending DTMF?
DTMFMode is rfc2833 on the Asterisk side and Auto on the phone side.

Thanks.

-- James

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