[Spce-user] multi-homed config

[Joel Smith | VOZELIA]

Hi, 

Regarding this scenario:

[Subscriber-PublicIP] -> [SPCE-WAN] [SPCE-LAN] -> [Asterisk] 

Say SPCE-LAN and Asterisk network is 192.168.1.0/24.


For an INVITE originated on Asterisk ending on the subscriber, does the 192.168.1.X IP of the Asterisk need to be able to reach the SPCE-WAN address?

Is it not enough with the Asterisk talking to SPCE-LAN?



Best regards,

----- Original Message -----
> From: "gerry kernan" <gerry.kernan at infinityit.ie>
> To: "Andrew Pogrebennyk" <apogrebennyk at sipwise.com>
> Cc: "Spce-user" <Spce-user at lists.sipwise.com>
> Sent: Friday, March 7, 2014 6:28:39 PM
> Subject: Re: [Spce-user] multi-homed config
> 
> 
> Hi Andrew
> 
> It was just the network location I was testing from , it had a
> Sonicwall firewall when I disable Nat option b2bua on the firewall
> all is ok
> 
> Thanks
> 
> Gerry
> 
> Sent from my iPhone
> 
> > On 7 Mar 2014, at 15:41, "Andrew Pogrebennyk"
> > <apogrebennyk at sipwise.com> wrote:
> > 
> > Hi Gerry,
> > 
> > this looks like a NAT problem because there is no response from
> > behind
> > NAT. Looks like the client doesn't receive the INVITE. It's
> > unlikely
> > that this is related to extra_sockets. Are all users having this
> > problem? Are all or most of them behind this particular NAT/router?
> > 
> > Check if the SPCE sends OPTIONS pings to the right port (from which
> > the registration was received, this is location.received column in
> > kamailio DB). Also check if the UA supports the NAT keep-alive
> > feature
> > and try enabling it.
> > 
> > Regards,
> > Andrew
> > 
> >> On 03/06/2014 04:36 PM, gerry kernan wrote:
> >> Hi
> >> 
> >> Apologies for more questions , but I seem to have broken something
> >> in sipwise. When I added the extra_sockets option and applied the
> >> config I haven't been able to route calls. If I try calling an
> >> extension it goes straight to VM. , sip trace below is incoming
> >> call to 35315549053 , the sip subscriber for this number is
> >> polycomvvx at 78.137.191.37.  This phone is connect via NAT from WAN
> >> address 82.141.241.25. and have an LAN address of 192.168.200.134.
> >> I don't see any enteries for this address in the sip debug. Would
> >> the change ive make cause this? I have remove the entery from
> >> config.yml and re-applied the config but still have the problem.
> >> 
> >> Mar  6 15:25:11 sipwise lb[18119]: NOTICE: <script>: Relaying
> >> request, du='sip:82.141.241.25:33179', fs='udp:78.137.191.37:5060'
> >> - R=sip:polycomvvx at 82.141.241.25:11254
> >> ID=oFfxx8rh.Z8Hy0 at 89.234.66.178
> >> 
> 
> 
> _______________________________________________
> Spce-user mailing list
> Spce-user at lists.sipwise.com
> http://lists.sipwise.com/listinfo/spce-user
>