[Asterisk-Dev] channels/chan_sip.c 1.677 and Polycom

[Andrew Lindh]

Still having problems with Polycom SIP phones since the chan_sip.c CVS update
to 1.677. This seems to be another polycom issue (xten and grandstream work).
It affectes CALLED polycom phones (from SIP or ZAP).

Phone 2 is the called polycom phone (with current 1.41 SIP firmware)...
Phone 1 (SIP or ZAP, any phone) calls phone 2 (the polycom)
If phone 1 (caller) ends the call first then the channel is closed and both
phones/channels end the call, all is good.
If phone 2 (called phone) ends the call then asterisk and phone 1 (caller)
leaves the channel open and does not end the call.

Asterisk gets the BYE but does not respond/ACK to it.

<-- SIP read from 204.213.176.210:5060: 
BYE sip:301 at 204.213.176.139 SIP/2.0
Via: SIP/2.0/UDP 204.213.176.210;branch=z9hG4bKc56cfdad7FC9E90C
From: <sip:310 at 204.213.176.210>;tag=8F8AA47A-37BB5B03
To: "Andrew" <sip:301 at 204.213.176.139>;tag=as4a0d6a8d
CSeq: 1 BYE
Call-ID: 7a415d7b468a077471727aff7c023a86 at 204.213.176.139
Contact: <sip:310 at 204.213.176.210>
User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.4.1
Max-Forwards: 70
Content-Length: 0

Asterisk terminates the channel after the no RTP timeout.

Once the call is answered no additional call functions work (at least
hold and transfer) from the polycom.

Odd thing is the reject key (when rining) on the called phone does work
before the call is answered....

If needed I'll open a bug report with a trace when I have a minute.



  Andrew