[OpenSIPS-Users] Call from Asterisk to Opensips
haloha201 at yahoo.com
Fri May 6 15:55:45 CEST 2011
first thing you should try to read the asterisk SIP TRUNK and here is the basic example and i think the problem is asterisk not opensips
and make sure to check the debug from asterisk and opensips, i think you will get the clues :D
--- On Thu, 5/5/11, Duong Manh Truong <ngoahotanglongbk at gmail.com> wrote:
From: Duong Manh Truong <ngoahotanglongbk at gmail.com>
Subject: [OpenSIPS-Users] Call from Asterisk to Opensips
To: "OpenSIPS users mailling list" <users at lists.opensips.org>
Date: Thursday, May 5, 2011, 9:53 PM
Hi all, I've created sip trunk on Asterisk and defined asterisk server ip on address table of opensips
Then, from extension of Opensips , i can dial out to pstn through Asterisk
Now, i want to route PSTN call to the extension but when Asterisk receive the call from PSTN and dial Opensips through the Sip Trunki always got the message in the asterisk's console:
Called to-opensips/1001 -- SIP/to-opensips-00000745 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0)
(1001 is the extension of Opensips)
Then the call hangs up.
Anyone got this problem ? please help me the way to deal with!
Thanks so much!
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