[OpenSIPS-Users] Fwd: opensips+asterisk call dropping in 20 seconds

[cesar.fiestas]

In my opinion the 20 sec drop call is due to a NAT issue, check your NAT
setup and or configuration

On Tue, Jul 7, 2009 at 1:13 PM, ram-2 (via Nabble) <
ml-user+92105-174774996 at n2.nabble.com<ml-user%2B92105-174774996 at n2.nabble.com>
> wrote:

>
> Hi
>
> In continuation with the subject
>
> when i call intiated from Opensips the call drop in 20seconds
>
> but when i register directly from * box i dont see the call drop even for
> 20-30min of talk
>
> any suggestions
>
> Ram
>
> On Tue, Jun 30, 2009 at 8:35 PM, ram <talk2ram at ...<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3220575&i=0>
> > wrote:
>
>>
>>
>>  On Tue, Jun 30, 2009 at 5:20 PM, Bogdan-Andrei Iancu <bogdan at ...<http://n2.nabble.com/user/SendEmail.jtp?type=node&node=3220575&i=1>
>> > wrote:
>>
>>> Hi Ram,
>>>
>>> I found your email on the Asterisk mailing list also ;)
>>>
>>> So, to answer here also: do you get any reply back from Asterisk ?
>>>
>>
>> Hi Bogdan
>>
>> thanks for the reply
>>
>> I have made a quick Fix, iam not sure how far its good.
>>
>> Just put coment in  secret , in the Asterisk
>> Additional_a2billing_sip.conf. rather doing twise  authentication.
>>
>>
>> But i have another problem here with the Dispatcher,
>> dispatcher sending calls round robin,
>>
>> 1 st call to 1st *
>> 2nd call to 2nd *
>> 3 call to 3rd *
>>
>> if 2nd Asterisk fails to respond still Dispatcher module sending calls to
>> 2nd asterisk
>>
>> how can i fix this issue with Dispatcher, if any one of * box not
>> reachable it should detect and send call to 3rd *
>>
>> if 2nd comes back in to network and live, it should send to 2nd *
>>
>> how can i achive this ?
>>
>> Ram
>>
>>
>>
>
>
>
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