[Spce-user] Private Address in Contact - SPCE as SBC

[Deon Vermeulen]

Hi Andreas


All logs and configs are actual copy paste.

Where should I check for this?

If I disabled Asterisk (apart from other functionality) why are calls still “routed” to voicebox?

Shouldn’t calls be routed directly to SEMS (SBC) from Kamailio?


Thank you 

Kind Regards


On Mar 31, 2014, at 9:56 AM, Andreas Granig <agranig at sipwise.com> wrote:

> Hi,
> 
> Is this a copy/paste error or is your voicebox entry in the dispatcher
> table really "sip:127.0.0.1:" without a port? It's supposed to be 5070,
> if I remember correctly.
> 
> Andreas
> 
> On 03/31/2014 08:01 AM, Deon Vermeulen wrote:
>> Good Morning
>> 
>> Any possibility for an update on this?
>> Much appreciated.
>> 
>> 
>> Thank you very much
>> 
>> 
>> Kind Regards
>> 
>> 
>> On Mar 28, 2014, at 6:53 PM, Deon Vermeulen <vermeulen.deon at gmail.com> wrote:
>> 
>>> Here is my network.yml
>>> 
>>> http://pastebin.com/iwzmKYUM
>>> 
>>> 
>>> Thank you for the assistance
>>> 
>>> 
>>> Kind Regards
>>> 
>>> 
>>> 
>>> On Mar 28, 2014, at 6:51 PM, Deon Vermeulen <vermeulen.deon at gmail.com> wrote:
>>> 
>>>> I forgot to mention that I disabled:
>>>> 
>>>> Asterisk
>>>> Rateomat
>>>> CSV Conference
>>>> 
>>>> Here is my config.yml
>>>> 
>>>> http://pastebin.com/nhFACU5w
>>>> 
>>>> 
>>>> Thank you
>>>> 
>>>> Kinds Regards
>>>> 
>>>> 
>>>> 
>>>> On Mar 28, 2014, at 6:44 PM, Deon Vermeulen <vermeulen.deon at gmail.com> wrote:
>>>> 
>>>>> Hi Andrew
>>>>> 
>>>>> Here within output:
>>>>> 
>>>>> +----+-------+--------------------+-------+----------+-------+---------------------+
>>>>> | id | setid | destination        | flags | priority | attrs | description         |
>>>>> +----+-------+--------------------+-------+----------+-------+---------------------+
>>>>> |  1 |     2 | sip:127.0.0.1:     |     0 |        0 |       | Voicemail servers   |
>>>>> |  2 |     3 | sip:127.0.0.1:5080 |     0 |        0 |       | Application servers |
>>>>> |  3 |     4 | sip:127.0.0.1:5090 |     0 |        0 |       | Fax2Mail servers    |
>>>>> +----+-------+--------------------+-------+----------+-------+---------------------+
>>>>> 
>>>>> Thank you for the assistance.
>>>>> 
>>>>> 
>>>>> Kind Regards
>>>>> 
>>>>> 
>>>>> 
>>>>> On Mar 28, 2014, at 3:30 PM, Andrew Pogrebennyk <apogrebennyk at sipwise.com> wrote:
>>>>> 
>>>>>> Hello,
>>>>>> 
>>>>>> in the log I see a message "Call from Voicebox". This can happen if the
>>>>>> source IP 105.188.0.1 and port 5061 is present in the dispatcher table,
>>>>>> e.g. if the system IP was changed to this IP at some point.. Could you
>>>>>> please share the output of:
>>>>>> mysql -e "select * from kamailio.dispatcher;"
>>>>>> ?
>>>>>> Thanks.
>>>>>> Andrew
>>>>>> 
>>>>>> On 03/28/2014 11:29 AM, Deon Vermeulen wrote:
>>>>>>> Good Day
>>>>>>> 
>>>>>>> I’ve setup spce 2.8 as an SBC based where interconnect is a subscriber
>>>>>>> and connection back to our internal proxy is a SIP Peer.
>>>>>>> 
>>>>>>> Based on concept
>>>>>>> from http://www.sipwise.com/news/technical/byov-system-spce-as-sbc/
>>>>>>> 
>>>>>>> Just a note that it is not setup with /“peer_auth_register”/
>>>>>>> 
>>>>>>> 
>>>>>>> Current system configuration is basic as per 2.8 manual.
>>>>>>> 
>>>>>>> Single domain  :  102.0.0.1
>>>>>>> 
>>>>>>> Default Billing with a Default route and system default “Free Default
>>>>>>> Zone” Fee.
>>>>>>> 
>>>>>>> Peer Trunk between internal proxy and spce is active.
>>>>>>> Peer Server has “force_outbound_calls_to_peer:” enabled.
>>>>>>> 
>>>>>>> There are No rewrite Rules.
>>>>>>> 
>>>>>>> Subscriber is configured as follows:
>>>>>>> -     Active Device Registrations:  *sip:*@105.188.0.1 *
>>>>>>> -     /force_outbound_calls_to_peer:/*enabled *
>>>>>>> -     /e164_to_ruri:/*enabled *
>>>>>>> -     /allow_out_foreign_domain:/*enabled *
>>>>>>> -     Trusted Sources: *105.188.0.1     UDP     .* *
>>>>>>> 
>>>>>>> I’ve pasted a SIP trace as well as an output of the proxy logs on pastebin:
>>>>>>> SIP Trace     :     http://pastebin.com/0qZSJ2K1
>>>>>>> Proxy Log     :     http://pastebin.com/aYnL9kUa
>>>>>>> 
>>>>>>> From what I can make from this is the 192.168.34.208 in the Contact.
>>>>>>> It looks like, according to the proxy logs, that it gets stripped or not
>>>>>>> recognised
>>>>>>> 
>>>>>>> [avpops_impl.c:327]: failed to parse uri
>>>>>>> Mar 26 16:34:03 sbc01 /usr/sbin/kamailio[2258]: INFO: <script>: Load
>>>>>>> dialplan IDs for domain 'sip:3727121266@'
>>>>>>> 
>>>>>>> Unless I’m wrong is there anyone that could help me out with this?
>>>>>>> 
>>>>>>> 
>>>>>>> Thanks
>>>>>>> 
>>>>>>> Kind Regards 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> Spce-user mailing list
>>>>>> Spce-user at lists.sipwise.com
>>>>>> http://lists.sipwise.com/listinfo/spce-user
>>>>> 
>>>> 
>>> 
>> 
>> 
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