[Spce-user] Network Configuration
Daniel Grotti
dgrotti at sipwise.com
Fri Apr 19 09:03:12 EDT 2013
So, in this case you need to check you Inbound Rewrite Rules for Calee
as the callee is 0044113001417 instead of 44113001417, as your alias:
No matching rewrite rules for '0044113001417' found
br,
Daniel
On 04/19/2013 02:55 PM, Deon Vermeulen wrote:
> 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
>>
>>
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
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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.
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>> 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
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>> 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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