On 30 Mar 2015, at 16:54, Mark Michelson <[email protected]> wrote:
> On 03/28/2015 08:06 PM, Joshua Colp wrote: >> George Joseph wrote: >>>> The fact that it goes to unavailable would be a bug. Why does it do so? >>> >>> Mark should probably chime in here but I think it's because the >>> earliest you could get a response from pjsip when a contact isn't >>> reachable is the unconfigurable 32 seconds. As I said that's a long >>> time to leave a contact available when it really isn't. Without >>> implementing our own timer setting the contact to unavailable was >>> probably the lesser of 2 evils. >> >> No matter what there's going to be a period where you are potentially wrong. >> I don't think making it unavailable was done on purpose. >> > Actually, I believe the timer may be configurable. In the type=system > settings, there are timer_t1 and timer_b settings. timer_t1 is the base used > for determining the retransmission interval, and timer_b is the maximum time > we will wait before giving up sending the request. The defaults for these > values are 500 ms and 32000 ms respectively. If you were to change timer_b to > be a smaller value, then presumably you would have a shorter time before the > transaction times out. > > A couple of caveats about these settings > 1) Since they're in the "type=system" settings, any change you make requires > an Asterisk restart in order to take effect. Are you serious? That's a very strange design. Without knowing anything about PJSIP, I think that is something that needs to be fixed. There are several SIP phones based on PJSIP where I can set timers without restarting. Wonder how they did it. > 2) PJSIP applies these timers globally. They will affect ALL SIP > transactions, not just the OPTIONS transactions from the qualify checks. That seems very strange, but I have to trust you here. It makes the channel driver rather unusable in gateway situations where it's a requirement that I have one set of timers for my internal systems and one for external clients. /O -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
