If two phones are registered with OpenSIPS and they call each other why
would you send the SIP messages to Asterisk? You need to set up route logic
so that if two local users call each other then the asterisk boxes are kept
out of the equation.
On , Schneur Rosenberg <[email protected]> wrote:
I think I see the problem but I don't know how to fix it, my system
works as follows, opensips is used as a registrar and as a load
balances, asterisk handeles everything besides the registration, so
when a call comes in from a DID it goes to opensips which sends it to
asterisk to dsecide what to do with the call, then if it needs to go
to a phone asterisk sends it back to opensips (because the phones are
registered to opensips) the problem is that when a BYE happens
Asterisk sends the BYE to Opensips and Opensips sends back a 404 Not
here. When I call from one phone on the system to another phone on the
same opensips, the phone sends a BYE to opensips which sends it to the
asterisk but the BYE never gets sent to the called phone.
Please help me forward the BYE to the phone, it seems to me that
Asterisk needs to send the BYE to the phone but does not do it.
S. Rosenberg
On Tue, Nov 15, 2011 at 12:49 AM, Schneur Rosenberg
[email protected]> wrote:
> 8
>
> On Tue, Nov 15, 2011 at 12:47 AM, [email protected]> wrote:
>> So if you show 177 dialogs but you know for a fact that there are
really
>> only 2 active calls then you have a lot of stale dialogs. I believe
you are
>> probably not doing anything with the BYE or CANCEL messages. So since
you
>> have 177 dialogs how many LB resources are currently being used?
>>
>>
>>
>> On , Schneur Rosenberg [email protected]> wrote:
>>> I will have to try that next time it happens, the problem is I hope
>>>
>>> there wont be a next time, I need it fixed before next time :-)
>>>
>>> the db mode is set to 1 "modparam("dialog", "db_mode", 1)"
>>>
>>>
>>>
>>> I checked my database I currently have 177 dialogs even though there
>>>
>>> is only 2 calls in the system, how do I set the system to release them
>>>
>>> when the call is finished?
>>>
>>>
>>>
>>> S. Rosenberg
>>>
>>> On Tue, Nov 15, 2011 at 12:21 AM, [email protected]> wrote:
>>>
>>> > What if you do the following when you think things are maxed out
>>>
>>> > "opensipsctl fifo dlg_list"
>>>
>>> >
>>>
>>> > Do any dialogs show up? How is your
modparam("dialog", "db_mode", ?) set
>>> > up?
>>>
>>> >
>>>
>>> > Just wondering if you have some stall dialogs and the LB module
still
>>> > thinks
>>>
>>> > there are active calls.
>>>
>>> >
>>>
>>> > Just guessing.
>>>
>>> >
>>>
>>> >
>>>
>>> >
>>>
>>> > On , Schneur Rosenberg [email protected]> wrote:
>>>
>>> >> I'm using opensips as a loadbalancer for 2 asterisk servers, almost
>>>
>>> >>
>>>
>>> >> every day the system stops accepting new calls and when I run
>>>
>>> >>
>>>
>>> >> "opensipsctl fifo lb_list" it says "Resource:: pstn max=50 load=50"
>>>
>>> >>
>>>
>>> >> for both servers, even though there is no calls on either system,
how
>>>
>>> >>
>>>
>>> >> can I know whats causing the system to think there are calls while
>>>
>>> >>
>>>
>>> >> there is no calls, and what can I do to fix it?
>>>
>>> >>
>>>
>>> >>
>>>
>>> >>
>>>
>>> >> when I do a restart on opensips it clears the calls.
>>>
>>> >>
>>>
>>> >>
>>>
>>> >>
>>>
>>> >> thanks
>>>
>>> >>
>>>
>>> >> S. Rosenberg
>>>
>>> >>
>>>
>>> >>
>>>
>>> >>
>>>
>>> >> _______________________________________________
>>>
>>> >>
>>>
>>> >> Users mailing list
>>>
>>> >>
>>>
>>> >> [email protected]
>>>
>>> >>
>>>
>>> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>> >>
>>>
>>> >>
>>>
>>> > _______________________________________________
>>>
>>> > Users mailing list
>>>
>>> > [email protected]
>>>
>>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>> >
>>>
>>> >
>>>
>>>
>>>
>>> _______________________________________________
>>>
>>> Users mailing list
>>>
>>> [email protected]
>>>
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>> _______________________________________________
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>>
>>
>
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