Hello,

In openser.cfg:
modparam("rr", "enable_full_lr", 1)
modparam("rr", "append_fromtag", 1)

if (!method=="REGISTER"){
                record_route();
};

if (loose_route()) {
                t_on_reply("1");
                if (!t_relay()) {
                   sl_reply_error();
               };
                exit;
};

if (method=="INVITE"){
             ds_select_dst("2", "0");
             #ds_select_dst("2", "4");
            #sl_send_reply("100","Trying");
             forward();
             exit;
};

When I make a call from Sip client registed with openser to PSTN through asterisk-b2bua, openser transfer Invite and 200  messages between Sip client and asterisk-b2bua ok. But when sip client or pstn side  hang-up openser do not forward BYE message to the correct destination (forward BYE to asterisk-b2bua or to sip client), therefore the call on other side do not disconnect.
Where is incorrect in openser.cfg?

Thanks and best regards

On 7/18/07, Daniel-Constantin Mierla < daniel@voice-system.ro> wrote:
Hello,

use record routing (see rr module) to ensure the right path of in-dialog
requests.

Cheers.
Daniel


On 07/17/07 05:19, Ha Noi Telecommunications wrote:
> Hi!
>
> I am using OpenSer with two Asterisk-b2bua
>
> Sip
> client<--------->OpenSer<--------------------->Asterisk-b2bua<------->PSTN
>                                 |
>                                 |
>
> <----------------------------->Asterisk-b2bua<----------->PSTN
>
> In OpenSer configure file  I am using  ds_select_dst("2", "4"); to
> perform load sharing the calls to PSTN.
> But when Sip client hang up first, I don't konw how to make OpenSer
> forward the Bye message from Sip client to correct Asterisk-b2bua to
> hang up the call at PSTN side.
>
> Can any body can help me.
>
> Thanks and best regards
>
> ------------------------------------------------------------------------
>
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