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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