sorry, I'm using the qualify option in asterisk, which i believe sends a
request over the SIP connection periodically.
http://www.voip-info.org/wiki/view/Asterisk+sip+qualify
Default is 60 seconds, I have it set to 15sec for that device.


On Mon, Oct 14, 2013 at 10:07 AM, Vick Khera <[email protected]> wrote:

> What do you mean pinged? Are they actually exercising the UDP connection
> for the SIP connection, or are you pinging them via some other connection?
>
>
> On Sun, Oct 13, 2013 at 6:44 AM, palesius . <[email protected]> wrote:
>
>> Already tried that, I think they are pinged every 30sec from the asterisk
>> side.
>>
>>
>> On Thu, Oct 10, 2013 at 10:05 AM, Vick Khera <[email protected]> wrote:
>>
>>> Can you configure your phones to use do a keepalive ping? It sounds like
>>> the states are timing out.
>>>
>>>
>>>
>>> On Wed, Oct 9, 2013 at 5:44 PM, palesius . <[email protected]> wrote:
>>>
>>>> To take a break from all the NSA talk...
>>>>
>>>> I'm having some trouble routing traffic over an openvpn tunnel between
>>>> two pfsense firewalls. Asterisk server on one end, a couple of different
>>>> phones on the other side.
>>>>
>>>> It was working fine when we had monowall on both ends. (W/ipsec tunnel)
>>>> Since changing to pfsense it will register with the server just fine
>>>> but will lose it's connection anywhere from a few minutes to hours later.
>>>>
>>>> I've tried both ipsec and openvpn tunnels and have pretty much the same
>>>> result. I know mono and pfsense use a diffrerent firewall engine, is there
>>>> something obvious I should set/change to fix this.
>>>>
>>>> I had kind of dropped the issue a few months ago but wanted to take
>>>> another stab at it. I'll try to do some packet captures but don't have any
>>>> at the moment. Just hoping there is some easy general fix for getting SIP
>>>> working that someone else has already discovered.
>>>>
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>>>>
>>>
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