[Spce-user] A little help please

Paul pasha at prosperity4ever.com
Mon Jul 22 20:01:40 EDT 2013


Hi Andrew,

Thanks so much for this last message, that was key to my whole problem. 
I messed up network.yml so I reinstalled sipwise CE and only applied 
the machine's IP as domain, created trunk from FS and everything is 
fine, I can route calls incoming and outgoing now just fine.

I am wondering if you can help me with one more issue that has been 
causing me to nearly pull my hair out. My calls (both incoming and 
outgoing) are being dropped after 30-31 seconds every single time.

Naturally I am suspecting NAT issues, I have tried everything I can 
think of in FreeSwitch config with no changes, so I want to look at 
NGCP (as this is my in and out gateway) to see if the problem might be 
there.

One thing I noticed is that under "registere devices" for the FS trunk 
in the NGCP GUI, it shows "Nat=Yes" this should not be the case because 
both freeswitch and ngcp are on the 10.0.0.0/24 network. Is there 
somewhere in the network configuration that I need to let NGCP know 
that FS (10.0.0.34) can connect to it without NAT because they are on 
the same subnet?

Thanks in advance!

Paul

On Sat, 13 Jul, 2013 at 3:54 AM, Andrew Pogrebennyk 
<apogrebennyk at sipwise.com> wrote:
> Hi,
> You should add 10.0.0.40 as a domain in NGCP panel, if this is the 
> address of your machine, and remove 10.0.0.34 (this will delete 
> subscribers provisioned in that domain too), because this is the FS 
> address. After this there should be some noticeable progress.
> 
> 
> Paul <pasha at prosperity4ever.com> wrote:
> 
> >On Fri, July 12, 2013 12:45 pm, Andrew Pogrebennyk wrote:
> >> Hi Paul,
> >>
> >>
> >> regarding the learning process, we have some diagrams to explain 
> the
> >> system architecture here:
> >http://sipwise.com/doc/2.8/spce/ar01s02.html,
> >> not sure if you've seen them. In short, every SIP message passes
> >through
> >> kamailio-lb which does basic DoS attack protection, and every 
> INVITE
> >goes
> >> from lb to proxy them sems, which establishes a new SIP dialog with
> >_b2b-1
> >> suffix in Call-ID and sens INVITE to lb for passing it finally to
> >callee..
> >
> >I will keep reading and rereading, thanks for the URL.
> >
> >>
> >>
> >> On 07/13/2013 04:31 AM, Paul wrote:
> >>
> >>> Andrew as always you are right :)
> >>>
> >>>
> >>> I adjusted a few more things in the trunk reg config, applied
> >settings
> >>> and now I have logs in the proxy log here they are:
> >>
> >> Yes, I've noticed the 500 error in the lb log and was already
> >expecting
> >> to see the "Dropping local branch" message in proxy log:
> >>
> >>> [...]
> >>> /usr/sbin/kamailio[11011]: INFO: <script>: Dropping local branch -
> >>> R=sip:gw+ngcp at 10.0.0.34:5080;transport=udp;gw=ngcp
> >>> ID=C550303D at 99.88.77.66
> >>>
> >>
> >> If PBX running on the same host by chance? Of have you added
> >10.0.0.34
> >> as Domain in admin panel, same as 10.0.0.40? I'm asking because 
> from
> >the
> >> log it looks like ngcp address is 10.0.0.40, the client is 
> registered
> >> behind 10.0.0.34, but ngcp detects that address as local so refuses
> >to
> >> forward the call there.
> >
> >PBX is not running on the same host.
> >
> >NGCP: 10.0.0.40
> >FreeSwitch: 10.0.0.34 (where call should end up)
> >
> >I did have 10.0.0.34 as a domain in NGCP GUI, but not 10.0.0.40, just
> >added it, see if it makes any difference.
> >
> >In domains I have --> 10.0.0.34
> >In Subscribers I have --> DID at 10.0.0.34
> >
> >In freeswitch I setup a trunk that connects as follows:
> >
> >user: DID at 10.0.0.34
> >pass: password
> >realm: 10.0.0.34
> >proxy: 10.0.0.40
> >
> >Both freeswitch and NGCP see the connection (freeswitch lists trunk 
> as
> >registered, NGCP shows it under active devices for that subscriber).
> >
> >Is there a link or document you can refer me to that explains how 
> NGCP
> >analyzier from from-header to pull out the user the DID is attached 
> to
> >and
> >then decides on how to forward it?
> >
> >Thanks
> >
> >Paul
> >
> >>
> >>>
> >>> Based on this it looks to me like the Call-ID (error in logs) is
> >what
> >>> terminates the call and never actually pushes it out to the
> >freeswitch
> >>> trunk (I don't see anything in the freeswitch logs at all about a
> >call
> >>> coming in etc, and I do have a route for "public" context that 
> will
> >>> catch all).
> >>>
> >>> Thanks!
> >>>
> >>>
> >>> Paul
> >>>
> >>
> >>
> 
> -- 
> Sent from my Android phone with K-9 Mail. Please excuse my brevity.
> -- 
> Sent from my Android phone with K-9 Mail. Please excuse my brevity.
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