[Asterisk-Users] Asterisk SIP sound issue

[Jeff Ramsey]

I have an asterisk box build from cvs stable that I am trying to use with 5
IP500 Polycom SIP phones. I can receive calls in through a digium wctdm
line. I can call out from the SIP IP500 phones to a PSTN number through the
same card. In other words, incoming and outgoing calls work just fine. It is
extension to extension calls that I have issues with.

When I call one SIP IP500 from the other, the call is connected, it is using
ulaw, but I cannot hear the other person, from either end, no matter who
makes the call. The only way I have found to make it work, it to put the
call on hold, (which makes the hold music come on, and I can hear that...)
and then pick the call back up. After picking the call back up, I can use
the call like normal. I can hear and be heard. I've checked with the
asterisk server, and the codec is ulaw the entire time the call is
connected.

I have an extension setup that plays back the date and time, and the sound
is fine on that extension, so I am really lost as to why this is happening.

Please help. I've been stuck here for days now.

-- 
Jeff Ramsey