[asterisk-users] problems with creating a call

[Peter den Hartog]

hehe i figured it out.. it was really stupid :)

i use opensips as an sip proxy, and i configured opensips to only react on
packages from local ip's.. asterisk was sending to an external ip and that
way i created my own little loop :), changed in sip.conf all the hosts to
the internal ip of opensips and it worked..

thanks for the input tho :)!

Peter

On Wed, Feb 10, 2010 at 2:44 PM, Kevin P. Fleming <kpfleming at digium.com>wrote:

> Peter den Hartog wrote:
> > Hello,
> >
> > I installed Asterisk in a linonde cloud debian 5, and i'm trying to
> > create a first call but when i try to set up the call i see the
> > following message:
> >
> >     -- Called 100 at 100
> >     -- Now forwarding SIP/105-00000008 to 'Local/100 at default' (thanks to
> > SIP/100-00000009)
> >     -- Executing [100 at default:1] Dial("Local/100 at default-c2a9;2",
> > "SIP/100 at 100") in new stack
> > [Feb 10 13:31:25] WARNING[3639]: app_dial.c:1712 dial_exec_full:
> > Skipping dialing interface 'SIP/100 at 100' again since it has already been
> > dialed
> >
> > i'm calling from 105 to 100 (100 is registred at another domain, defined
> > in sip.conf that's why there is an 100 at 100)
>
> The device at SIP/100 sent a redirect (forward) message back to Asterisk
> suggesting that the call be sent to extension '100'. Asterisk refuses to
> call that device again because it's already been called in that
> particular instance of Dial and doing so would just result in an
> infinite loop.
>
> You need to figure out why the device at SIP/100 told Asterisk to
> forward the call when you were expecting it to just accept it.
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> skype: kpfleming | jabber: kpfleming at digium.com
> Check us out at www.digium.com & www.asterisk.org
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Groet // Kind regards,
Peter den Hartog
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100210/c10ec058/attachment.htm