[Spce-user] sipwise as transit switch

Andrew Pogrebennyk apogrebennyk at sipwise.com
Mon Apr 30 05:57:28 EDT 2012

Hello William,
If I understood you correctly you need to enable peer relay (search for
option allow_peer_relay in /etc/ngcp-config/config.yml, or enable
force_outbound_calls_to_peer preference for the inbound peering). If
that doesn't work I'd like to see kamailio-proxy.log.

On 04/30/2012 11:43 AM, William Ikiabo wrote:
> Hi,
> I want to thank the developers of SIPWISE for the wonderful product
> including every member of this forum for providing support in no small
> measure.
> We've been using VoIPSwitch for some a long time now as our transit
> switch between our Huawei Csoftx3000 and our interconnection carrier,
> There is a 1:1 NAT between SIPWISE 2.5 and the Internet, I have edited
> the config.yml for appropriately and every thing seems to work, NAT
> seems not to be an issue.
> What I intend to achieve is to is this, when subscribers on our mobile
> network dial numbers like 0703, 0803, 0810 which already hits sipswise
> should be switched to our interconnect carrier.
> In the opposite direction when subscribers on other networks dial 0895XX
> XXX, 2348950X XXX which already hits sipwise it should be switched to
> our MSC .
> The problem is when these calls hit sipswise it checks for the numbers
> locally and sends 404 Not Found
> I created a domain on sipwise and tried to terminate calls on the MSC
> and it's working in one direction. but have not been able to terminate
> calls from the MSC to the PSTN through sipwise.
> I am using the default rewrite rules for the two SIP peers, I think the
> rewrite rules are ok, I suspect the problem is with the caller pattern.
> Please I need your help to help resolve this issue.
> Warm Regards

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