[asterisk-users] Muted sound on a Linksys 962

[James Lamanna]

> Date: Tue, 27 Jan 2009 12:50:36 -0600
> From: "Danny Nicholas" <danny at debsinc.com>
> Subject: Re: [asterisk-users] Muted sound on a Linksys 962
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
>        <asterisk-users at lists.digium.com>
> Message-ID: <A8D2EF23F63B4A42B264BB151C5D65DB at db0005>
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>
> 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.

I'm not sure how that would help in this case.
The call is answered by the remote end and then
the caller can hear the audio of the IVR menus.
Or am I missing something here?

-- James

>
> -----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