[Spce-user] calls not going thru

Andreas Granig agranig at sipwise.com
Wed Dec 29 09:17:59 EST 2010

On 12/29/2010 02:47 PM, Suman Gandham wrote:
> I removed the caller and callee and made it empty but its same . 
> What I am trying to achieve is when a sip client dial a number it should be route to the SIP peer VPS which i have only one configured. 
> for example all my clients dial with 00 or + and this dial plan should be routed to the gateway VPS which is configured . 

If your clients dial with 00 or +, then you need a domain rewrite rule
for callee as noted in chapter of
http://sipwise.com/doc/spce/ar01s03.html#_creating_domains , so it gets
normalized correctly. Have you configured these rules?

If that doesn't help either, please send me the log entries in
/var/log/ngcp/kamailio.log for such a call.


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