[asterisk-users] Realtime peers and trunks coming from the same IP

[Ricardo Carvalho]

Thanks Sammy, I think I'll stop using SIP realtime.

Regards,
Ricardo.



On Mon, May 21, 2012 at 5:14 AM, SamyGo <govoiper at gmail.com> wrote:

> Hello Ricardo,
> The reason why your asterisk refused the calls from phone registering on
> SIP proxy is that it only gets INVITE of the call from: a user that is
> defined BUT Not Registered within asterisk.
> The easy way of solving this is
> 1- Stop asterisk SIP realtime and let only the SIP proxy handle
> registrations.
> 2- Tell asterisk to accept calls from the SIP proxy only (create a SIP
> peer for proxy)
> This will make everything work.
>
> Regards,
> Sammy.
>
>  On Sat, May 19, 2012 at 9:15 PM, Ricardo Carvalho <
> rjcarvalho.lists at gmail.com> wrote:
>
>>  I use an SBC to protect my pool of asterisk servers and as trunking
>> endpoint with SIP Telcos. Now I'm trying to implement SIP phone
>> registration to be delegated through the SBC, as a proxy.
>>
>> It doesn't work. It just works when I don't use realtime peers at the
>> asterisk servers. Using realtime SIP peers, since there is one SIP phone
>> that gets his registration delegated through the SBC, any inbound call that
>> tries to reach any asterisk server, coming from any SIP Telco trunk ended
>> at my SBC, gets refused in asterisk. As asterisk records the IP of the SBC
>> as the IP of the phone that has been registered, it "thinks" that those
>> calls coming from the SBC are calls coming from that phone, and it refuses
>> them with "401 Unauthorized" replies. I'm using asterisk 1.8.11.
>>
>> How can I surpass this problem? Is there any configuration that I'm
>> lacking on, or is this a limitation of asterisk?
>>
>> Thanks,
>> Ricardo.
>>
>> --
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>
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