[asterisk-dev] RTP streams suddenly stop

[Tony Mountifield]

In article <B1CA0D903B84954E83707132AA3A1C4906C1A09511 at mse23be3.mse23.exchange.ms>,
Jamuel Starkey <jamuel at hcvoip.com> wrote:
> Does dmesg or /var/log/messages show anything notable during that 5-min interval?

No, not at all. I'm certain it's something within Asterisk. See my other
recent posting for my current theory.

Cheers
Tony

> ----- 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 08:39:25 2010
> Subject: [asterisk-dev] RTP streams suddenly stop
> 
> I'm posting this to asterisk-dev because I am certain that I will need
> to dive into the code to identify and fix this problem. However, I'm not
> yet sure where to look, so would be grateful for some ideas!
> 
> The system in question talks SIP to an ITSP and is installed on their
> LAN in a colocated rack. It also has a couple of SIP phones dial into it
> over the Internet.
> 
> Every so often (once every week or two), the users complain of being cut
> off, and when dialling back in cannot hear any audio. This persists for
> several minutes before magically fixing itself.

-- 
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org