23 Jul
2007
23 Jul
'07
12:43 a.m.
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(a)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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