[Spce-user] Network Configuration
Deon Vermeulen
vermeulen.deon at gmail.com
Fri Apr 19 08:55:57 EDT 2013
Hi Daniel
Thank you for coming back to me.
Perhaps I'm not grasping this concept correct, but I don't have any
Peering Rules as the Carrier is a Subscriber on the 10.222.0.250 domain
with an Alias of 44113001417.
I have an internal Network with domain 192.168.0.250 and a Carrier with
a dummy/fake subscriber on domain 10.222.0.250.
The spce listens on 10.222.0.250 for the Carrier and on 192.168.0.250
for our internal network.
Is there any setting within the Carrier subscriber preferences I should
look at, or should the carrier be within the same domain as my internal
network for this to work?
Thanks for the assistance
Kind Regards
Deon
> Daniel Grotti <mailto:dgrotti at sipwise.com>
> April 19, 2013 1:02 PM
> Hi Deon,
>
> The call fails to select a proper PSTN gateway for:
>
> caller: sip:44839999989 at 192.168.0.250
>
> callee: sip:0044113001417 at 192.168.0.250;transport=udp
>
>
> It looks like you don't have any eligible peering rule for this
> callee/caller.
> How your Peering rules looks like ?
>
> br,
> Daniel
>
>
>
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> Deon Vermeulen <mailto:vermeulen.deon at gmail.com>
> April 19, 2013 12:51 PM
> I think I might have resolved the issue to run media-proxy on
> multiple interfaces on the same machine.
> Once I can confirm this is working will I elaborate on how I achieved
> this.
>
> I'm facing a problem though and it is as follows:
>
> I'm registered via SIP Client on my Internal Network
> 0839999989 at 192.168.0.250.
>
> One of the Carriers is registered to fake subscriber
> 0839999999 at 10.222.0.250 with Alias 44113001417.
>
> I've made 100% sure that my re-write rule is set on both the subscribers.
>
> I make call to 0044113001417 , but call fail with " No PSTN gateways
> available " in the proxy log.
>
> I'm also not 100% sure why it is not calling to the local hosted
> domain 10.222.0.250?
>
> The LB is listening on 10.222.0.250:6090.
>
> Attached the output of the call within the proxy log.
>
>
> Thanks again for any assistance.
>
> Kind Regards
> Deon
> Jon Bonilla (Manwe) <mailto:jbonilla at sipwise.com>
> April 18, 2013 11:14 PM
> El Thu, 18 Apr 2013 14:04:56 +0200
>
>
> I have found this requirement, where the peer requires a vpn or any other
> tunnel scenario to be established to send the traffic and they will
> only accept
> traffic from the tunnel. Most common carrier asking for this is
> British Telecom
> (BT)
>
> The extra_socket option does not work here because the rtp will still
> be sent
> and received in the default ip address, which is not the one used for the
> tunnel.
>
> 2 ways of dealing with it:
>
> * Create the tunnel in your router. This is the "righ way" (imho).
> I've also
> seen operator purchasing a small cisco router just for this function. It's
> easy and cheap.
>
> * Use an aditional spce to route the calls between the tunnel and the main
> spce. This is the same Thilo achieves with his kamailio-sems combo. This
> additional spce will listen in the tunnel address and that ip will be
> routeable from/to the main spce's default ip address.
>
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> Thilo Bangert <mailto:thilo.bangert at gmail.com>
> April 18, 2013 1:04 PM
> On Thursday, April 18, 2013 11:58:02 AM Deon Vermeulen wrote:
>> Hi
>>
>> Can anyone please help me to achieve the following network configuration:
>>
>> PROVIDER 2
>>
>>
>> SIP TRUNK
>>
>>
>> Internal Network ---- SIP TRUNK ----- SPCE ----- SIP TRUNK -----
>> PROVIDER 1
>>
>>
>> SIP TRUNK
>>
>>
>> PROVIDER 3
>>
>
> we have solved this by having provider specific peering gateways, consisting
> of a kamailio and sems combo on a different (virtual) host.
>
> kind regards
> Thilo
>
>
>> Thanks a lot.
>>
>> Kind Regards
>>
>> Deon
>
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> Spce-user at lists.sipwise.com
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> Deon Vermeulen <mailto:vermeulen.deon at gmail.com>
> April 18, 2013 11:58 AM
> Hi
>
> Can anyone please help me to achieve the following network configuration:
>
> PROVIDER 2
> |
> |
> SIP TRUNK
> |
> |
> Internal Network ---- SIP TRUNK ----- SPCE ----- SIP TRUNK -----
> PROVIDER 1
> |
> |
> SIP TRUNK
> |
> |
> PROVIDER 3
>
>
> Network Addresses and Listening Ports:
>
> Internal Network
> - Eth0
> ip: 192.168.0.10
> netmask: 255.255.255.0
> port: 5060
> protocol: udp
> rtp_ext
> web_int
> web_ext
>
> PROVIDER 1
> - Eth1
> ip: 172.16.1.10
> netmask: 255.255.255.0
> port: 6080
> protocol: udp
> rtp_ext
>
> PROVIDER 2
> - Eth2
> ip: 172.16.2.10
> netmask: 255.255.255.0
> port: 6085
> protocol: udp
> rtp_ext:
>
> PROVIDER 3
> - Eth3
> ip: 172.16.3.10
> netmask: 255.255.255.0
> port: 6090
> protocol: udp
> rtp_ext
>
>
> OBJECTIVES:
> 1. All services should be available on the Internal Network.
> 2. Only SIP and RTP services should be available on the PROVIDER networks.
> 3. Only the assigned SIP Ports should be available to the respective
> PROVIDER, ie. PROVIDER 1 only port 6080 udp.
> 4. I'm not sure how the roles should be setup for each respective
> interface?
>
>
> Thanks a lot.
>
> Kind Regards
>
> Deon
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