asterisk users - Oct 2004 - SPA-3k outbound calls...

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

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20041009/be744e0c/attachment.htm

On Sat, 9 Oct 2004 23:41:41 -0400, Jeff owen <owenj@surfree.net>
wrote:
> Ok, now since I have inbound working properly the outbound seemed to have
> gotten hosed.
What's the version of the firmware of your SPA3K? What are the PSTN
line settings, in particular  the VoIP-to-PSTN-gateway section and the
SPA's dialplan if you used any?

Finally, what does your SIP debug output on the Asterisk console say
when you try to dial out?

rgds
benjk

-- 
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.

NB: Spam filters in place. Messages unrelated to the * mailing lists
may get trashed.

On Sat, Oct 09, 2004 at 11:41:41PM -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)
FWIW I was fighting with this today and had to make this line like:

exten => _9NXXXXXXXXX,1,Dial(SIP/${EXTEN:1}@ipaddress_of_sipura:5061)

Maybe if the host was specified in sip.conf rather than being listed as
dynamic this wouldn't be necessary.

-- 
Ray

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
> 
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Mike Benoit <ipso@snappymail.ca>