[Asterisk-Dev] Issues with asterisk/channels chan_sip.c, 1.591, 1.592...

[Andrew Lindh]

After this patch my Polycom phones don't work anymore. Calls from polycom
to others don't work correctly. It gets worse in later patches.

Anyone else have this problem? Does anyone test with the Polycom Soundpoint IP?
I know cisco and cheap phones are popular, but the polycoms are nice and
not that much...

chan_sip.c version 1.591 works as expected.
1.592 the call rings and can be answered but there is no audio (either way)
1.593 no audio and when the polycom hangs up asterisk reports:
	Got SIP response 400 "Bad Request" back from <phone IP>
And the remote phone never gets word that the call is disconnected.

Current versions have the same results.

I can open a bug on this with more info, but figured I'd ask here first.

 -- Andrew


>From: markster at lists.digium.com
>To: asterisk-cvs at lists.digium.com
>Date: Sat, 18 Dec 2004 09:30:13 -0600 (CST)
>Subject: [Asterisk-cvs] asterisk/channels chan_sip.c,1.591,1.592
>
>Update of /usr/cvsroot/asterisk/channels
>In directory mongoose.digium.com:/tmp/cvs-serv18482/channels
>
>Modified Files:
>	chan_sip.c 
>Log Message:
>Merge olle's amazing ACK fix (bug #2687)
>