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:[EMAIL PROTECTED];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:[EMAIL PROTECTED] SIP/2.0
P-BYE sip:[EMAIL PROTECTED] SIP/2.0
A-SIP/2.0 200 OK

The failing SIP trace looks like this:

P-INVITE sip:[EMAIL PROTECTED];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:[EMAIL PROTECTED] SIP/2.0
P-BYE sip:[EMAIL PROTECTED] 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:[EMAIL PROTECTED]>;tag=B81927F5-872A14B0
To: <sip:[EMAIL PROTECTED];user=phone>;tag=as4c95cddd
Call-ID: [EMAIL PROTECTED]
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
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?
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