[OpenSIPS-Users] Call from Asterisk to Opensips

Mark Sayer datapipes at avtb.co.nz
Fri May 6 06:06:38 CEST 2011

You have provided us with the error message from Asterisk but what
have you looked to see what OpenSIPS is doing? Is ext1001 currently
registered with OpenSIPS? There are a number of ways that Asterisk and
OpenSIPS might be configured to operate together. You will have to
give us more information on your setup.


On Fri, May 6, 2011 at 1:53 PM, Duong Manh Truong
<ngoahotanglongbk at gmail.com> wrote:
> 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 Trunk
> i 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!
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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