[asterisk-dev] 900 second call drop - 420 Bad Extension error w/ SIP, IOS GW

[Jared Mauch]

On Mon, Dec 21, 2009 at 09:48:17AM -0600, Kevin P. Fleming wrote:
> Jared Mauch wrote:
> 
> >     -- Got SIP response 420 "Bad Extension" back from x.x.2.115
> >   == Spawn extension (verio-int, 600, 4) exited non-zero on 'SIP/dns.name.hidden-00000064'
> > 
> > 
> > The actual message received is usually something like this:
> > 
> > SIP/2.0 420 Bad Extension
> > Via: SIP/2.0/UDP x.x.27.16:5060;branch=z9hG4bK21db1330;rport
> > From: <sip:2149151350 at x.x.27.16>;tag=as22d8cfaf
> > To: <sip:2149151356 at x.x.2.115>;tag=6E691D88-4A
> > Call-ID: C8B6CF60-D86311DE-93A08FDB-20B8ACA9 at x.x.2.115
> > CSeq: 102 INVITE
> > Unsupported: timer
> > Content-Length: 0
> 
> Asterisk is trying to use SIP session-timers for this dialog, presumably
> because the other end offered support for them in the initial INVITE.
> However, when Asterisk sent a session refresh because the timer was
> about to expire, the other end rejected it.

	Sounds like it might be related to this change 
(1.6.1.4/1.6.1.6 chan_sip diff)

        }
 
        /* Session-Timers */
-       if (p->sipoptions == SIP_OPT_TIMER) {
+       if (p->sipoptions & SIP_OPT_TIMER) {
                /* The UAC has requested session-timers for this session. Negotiate
                the session refresh interval and who will be the refresher */
                ast_debug(2, "Incoming INVITE with 'timer' option enabled\n");


	I wonder if it's an IOS bug after all.

	Will do more testing later when the pbx is more idle.  This week
should be a good one for testing since there is less critical stuff going
on :)

	Thanks.

	- Jared

-- 
Jared Mauch  | pgp key available via finger from jared at puck.nether.net
clue++;      | http://puck.nether.net/~jared/  My statements are only mine.