[Asterisk-Users] SPA-3k outbound calls...

[Mike Benoit]

I submitted a bug report (http://bugs.digium.com/bug_view_page.php?
bug_id=0002620) regarding this issue a couple days ago, and it has since
been fixed. You can download the patch from the above link, or wait a
bit and it will probably be applied to the stable CVS branch.



On Sat, 2004-10-09 at 23:41 -0400, Jeff owen wrote:
> Ok, now since I have inbound working properly the outbound seemed to
> have gotten hosed.
> 
>  
> 
> In the extensions.conf I have it setup as:
> 
>  
> 
> exten => _9NXXXXXXXXX,1,Dial(SIP/${EXTEN:1}@pstn)
> 
>  
> 
> In the sip.conf I have it setup as:
> 
>  
> 
> [pstn] SPA-3k PSTN Line
> 
> type=friend
> 
> context=default
> 
> secret=supersecretpassword
> 
> port=5061
> 
> host=dynamic
> 
> dtmfmode=rfc2833
> 
> canreinvite=no
> 
> nat=no
> 
>  
> 
> Which should be correct for inbound and outbound calling, right?
> 
>  
> 
> However all I get when I try and dial out is another dial tone and if
> I try to dial a number a second time the call will go thru.  Kind of
> like dialing 98145551212, getting dial tone, then dialing 8145551212
> and the call gets connected then.
> 
>  
> 
> I’m not sure what needs to be set on the SPA-3k to allow calls to be
> made by what is passed to it.  However, when I look at the SIP debug I
> don’t even see the number listed as passed to the SPA-3k.
> 
>  
> 
> Any ideas on where to look or what to set?
> 
>  
> 
> Thanks,
> 
>  
> 
> Jeff
> 
> 
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-- 
Mike Benoit <ipso at snappymail.ca>