[OpenSIPS-Users] BYE request for proper signalling

[Bogdan-Andrei Iancu]

Denis,

in this case, are the other proxies involved in the call doing Record 
Routing ? if so, opensips dialog module take them into consideration 
when sending the BYE.

Regards,
Bogdan

Denis Putyato wrote:
> Hello Bogdan
>
> " because of some NAT presence, right ?"
>
> No, I need use IP address when there is more than one SIP proxy in call path. 
>
> -----Original Message-----
> From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
> Sent: Wednesday, February 02, 2011 3:36 PM
> To: OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] BYE request for proper signalling
>
> Hi Denis,
>
>  From SIP point of view, the BYE must be sent to the contact URIs . I 
> guess your contact is different than the layer3 IP because of some NAT 
> presence, right ? if so, use fix_nated_contact() for INVITE and 200 OK, 
> so that the received contact will be "fixed" with the layer3 IP, so the 
> dialog module will use the contact with a useful info.
>
> Regards,
> Bogdan
>
> Denis Putyato wrote:
>   
>> Hello!
>>
>>  
>>
>> I am using dialog module for control of call duration.
>>
>> When timeout of dialog expires I need Opensips send BYE not to caller 
>> and callee contact (which is stored during creation of dialog) but to 
>> IP address and port from which INVITE (caller) and 200 OK (callee) had 
>> been received.
>>
>>  
>>
>> Thank you for any help
>>
>>  
>>
>>  
>>
>> ------------------------------------------------------------------------
>>
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>
>
>   


-- 
Bogdan-Andrei Iancu
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