[asterisk-dev] [regression] Anonymous SIP calls are hung up after 900 seconds (Asterisk v. 1.6.0.18)

[Jamuel Starkey]

Check your SIP session-timers setting in [general]. And of course look at a pcap or ngrep of the signalling and pay particular attention to the session-timer that's negotiated at call setup (inital INVITE) and then again at around 900 seconds. 

Without an accurate view of the signaling its all guesswork. But you'll probably see a re-INVITE attempted that causes the call to get dropped (SIP 420 Bad Extension). 

Cheers,

JPS

----- Original Message -----
From: asterisk-dev-bounces at lists.digium.com <asterisk-dev-bounces at lists.digium.com>
To: asterisk-dev at lists.digium.com <asterisk-dev at lists.digium.com>
Sent: Thu Feb 04 09:39:53 2010
Subject: Re: [asterisk-dev] [regression] Anonymous SIP calls are hung up	after 900 seconds (Asterisk v. 1.6.0.18)

On Thursday 07 January 2010 20:54:55 Hans Petter Selasky wrote:
> Hi,
> 
> I have a problem where anonymous SIP calls, I.E. SIP calls without a
> registered username, start hanging up around 900 seconds (exactly).

I can confirm that this problem is fixed as of Asterisk v1.6.0.21.

Thanks!

--HPS

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