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-boun...@lists.opensips.org
[mailto:users-boun...@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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