[Spce-user] calls not going thru
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 18.104.22.168 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.
-------------- next part --------------
A non-text attachment was scrubbed...
Size: 900 bytes
Desc: OpenPGP digital signature
More information about the Spce-user