[asterisk-dev] 900 second call drop - 420 Bad Extension error w/ SIP, IOS GW
Olle E. Johansson
oej at edvina.net
Mon Dec 21 09:41:49 CST 2009
21 dec 2009 kl. 16.36 skrev Jared Mauch:
> I've been trying to track down an odd issue I've seen in versions
> after 22.214.171.124 (including 1.6.2).
> We have an IOS gateway that connects us to the PSTN for performing SIP
> inbound/outbound termination.
> What we see is when a call originates on the PSTN (so comes in the PRI
> and is handed to Asterisk) after 900 seconds the call is dropped.
> I'm not sure what is triggering this exactly. I've looked at SIP traces and
> getting the exact nuance sorted out is tough, but what I see is something
> similar to this:
> -- 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
This is not really a question about asterisk source code development yet, while it may become. Please use the asterisk-users list for issues you have with running asterisk.
The device is telling you that it doesn't support session timers. Check your setting for session timers in sip.conf. If you've turned it off and it still happens, please file a bug report.
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