[asterisk-dev] Enable SIP keepalive (OPTIONS?) for calls from unregistered but authenticated peers

[Johan Sandgren]

Yes ok, I'll try to enable these session timers and see how it works out.

Thanks guys

-----Ursprungligt meddelande-----
Från: asterisk-dev-bounces at lists.digium.com [mailto:asterisk-dev-bounces at lists.digium.com] För Olle E. Johansson
Skickat: den 3 juni 2013 15:44
Till: Asterisk Developers Mailing List
Kopia: Olle E. Johansson
Ämne: Re: [asterisk-dev] Enable SIP keepalive (OPTIONS?) for calls from unregistered but authenticated peers


3 jun 2013 kl. 11:59 skrev Walter Doekes <walter+asterisk-dev at osso.nl>:

> On 03/06/13 11:53, Johan Sandgren wrote:
>> I need activate SIP keepalives "OPTIONS"-packet (over UDP) for unregistered, but authenticated calls into asterisk.
>> I have played with the sip.conf settings but have not been able to enable any SIP keepalives when not registered when calling with SIP into Asterisk.
>> So I decided to go into the source code to modify it to send then even for unregistered calls.
> 
> Assuming you want this to keep NAT holes open, can't you use session-timers with a low expiry for this purpose?
> 
The OPTIONs are not related to "calls", Johan. It's just related to peers.

As Walter said, session timers are activated on a per call basis.


/O
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