[Asterisk-Users] SIP ringback problem with Polycom phones and CVS HEAD

[Kevin P. Fleming]

For the past week or two, our customers who have Polycom phones have 
been experiencing a problem... but our customers with Cisco phones do 
not have this problem.

The phones in question are:

Polycom SoundPoint IP300 (firmware 1.3.1 or 1.3.4)
Polycom SoundPoint IP500 (firmware 1.3.1 or 1.3.4)
Cisco 7960 (firmware 7.2 or 7.3)

The problem is this: when our Polycom users dial _some_ PSTN numbers, 
they hear one cycle of "ringback", then it's gone. However, the call is 
still proceeding, and if they wait for it to be answered the call 
proceeds normally (audio flows in both directions). When they dial 
_most_ PSTN numbers, this does not happen.

In fact, the calls are all following the same path: from the Asterisk 
server that the phones register to, over IAX to another Asterisk server, 
then out a PRI (these are all local calls).

I have run a "sip debug" trace of the successful and failing calls, and 
everything looks normal; there is only one difference for the failing calls.

The successful SIP trace looks like this (P-Polycom Phone, A-Asterisk):

P-INVITE sip:96027414660 at test.starnetworks.us;user=phone SIP/2.0
A-SIP/2.0 100 Trying
A-SIP/2.0 183 Session Progress
A-SIP/2.0 200 OK
P-ACK sip:96027414660 at 67.137.151.151 SIP/2.0
P-BYE sip:96027414660 at 67.137.151.151 SIP/2.0
A-SIP/2.0 200 OK

The failing SIP trace looks like this:

P-INVITE sip:96027414660 at test.starnetworks.us;user=phone SIP/2.0
A-SIP/2.0 100 Trying
A-SIP/2.0 183 Session Progress
A=SIP/2.0 180 Ringing
A-SIP/2.0 200 OK
P-ACK sip:96027414660 at 67.137.151.151 SIP/2.0
P-BYE sip:96027414660 at 67.137.151.151 SIP/2.0
A-SIP/2.0 200 OK

Note the additional "180 Ringing" message in this trace. When the 
Polycom phone receives this, it stops generating (or passing) ringback 
to the caller.

The actual message is this (but my email client has wrapped it):

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 
192.168.1.121;branch=z9hG4bK2ec810b2FB407A27;received=68.14.253.125;rport=1172
From: "3011" 
<sip:starnetworks.004575 at test.starnetworks.us>;tag=B81927F5-872A14B0
To: <sip:96233867319 at test.starnetworks.us;user=phone>;tag=as4c95cddd
Call-ID: 4a32ad31-c93f29a3-5a78678e at 192.168.1.121
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:96233867319 at 67.137.151.151>
Content-Length: 0


This was working fine before a recent upgrade to Asterisk; I believe it 
started being a problem after I upgraded to CVS HEAD from around 2004-12-10.

I assume this difference in the call trace is due to some difference in 
the call path through the PSTN (one path reports in-band progress, the 
other out-of-band, or something like that), but I don't understand why 
the phone would stop ringback when it receives this message. As it 
stands right now, I'm going to have to suppress these messages 
completely, as it's not a pleasant problem for my customers to deal with...

Anyone have any idea why this message would cause this problem, or what 
may have changed in chan_sip recently that might have changed the 
behavior in this area?