[OpenSIPS-Users] Call from Asterisk to Opensips

Brett Nemeroff brett at nemeroff.com
Fri May 6 15:04:40 CEST 2011

On Thu, May 5, 2011 at 10:53 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
Remember, just because you can dial out, doesn't mean that you are properly
registered. Also, depending on your configuration, there are many ways you
could be routing this call. I assume you are expecting the call to be routed
to the registered device. Have you checked it's registration status with:
opensipsctl ul show

If it is registered, then the problem is in your configuration and we
wouldn't really be able to help you out much without seeing what exactly
you've got setup.
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