[asterisk-users] Dynamically change sip.conf properties.

[Simon Elliston Ball]

Realtime only needs a sip reload if you are using static realtime, if  
you use the sippeers realtime it works just fine. See http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip

Note that the settings change will only take effect when your client  
re-registers, so you may want to set a reasonably low qualify value.

Simon


Simon Elliston Ball
simon at simonellistonball.com



On 11 Dec 2007, at 15:15, asterisk wrote:

> I don't know of a way without reloading. Realtime still needs a sip
> reload.
>
> Look at the dial command.   There are options that you can add that  
> will
> disable re-invites per call.
>
> Doug Gillespie
>
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Alex
> Balashov
> Sent: Monday, December 10, 2007 12:40 PM
> To: asterisk-users at lists.digium.com
> Subject: [asterisk-users] Dynamically change sip.conf properties.
>
>
> Is there a way to dynamically alter the sip.conf properties of a SIP
> peer
> in runtime without doing a SIP reload?
>
> I am specifically thinking of enabling reinvites for users dynamically
> based on whether they are registered from a public address.
>
> --
> Alex Balashov
> Evariste Systems
> Web    : http://www.evaristesys.com/
> Tel    : +1-678-954-0670
> Direct : +1-678-954-0671
>
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