[OpenSIPS-Users] opensips and asterisk

[David J.]

Sorry, The way I recommend doing this was assuming the user on the 
Asterisk box needed to be publicly reachable from anywhere.

I think that approach makes sense when using DID's and inbound routing 
that does need authentication.



On 5/4/10 12:55 PM, Olle E. Johansson wrote:
> 4 maj 2010 kl. 18.30 skrev Brett Nemeroff:
>
>    
>> Carmelo,
>> If you have an SIP peer that matches the host and port of the opensips server.. ie:
>> [opensips]
>> type=friend
>> host=<ip of opensips.
>> port=<port of opensips>  (can be omitted if port 5060)
>>
>> Then it'll match that.. typically if it's coming from opensips you'll want to add:
>> insecure=invite
>>
>> so that opensips won't be challenged to authenticate. Also be sure there is no secret set.
>>
>> I personally wouldn't do this using the default context as the other posters had recommended as that will allow *anyone* to send traffic to your asterisk server. Which I don't believe is what you really want to do. Instead, create a peer that is limited by IP and PORT allowed to send invites without a secret.
>>
>> Also be sure that the context for that peer is set to the right context and that if from the asterisk CLI you type:
>> dialplan show<RURI username>@<opensips context>
>> that it matches something you'd expect.
>>
>> On another note, are you performing a consume credentials? I think it *might* be possible that opensips is forwarding your UAC's credentials on to Asterisk if you are not..
>>
>>      
> If you want to ONLY match on IP/port, you need to use "type=peer".
>
> regards,
> /O
>
>    
>> -Brett
>>
>>
>> On Tue, May 4, 2010 at 8:02 AM, wüber<leone81 at gmail.com>  wrote:
>>
>> Hi Bogdan,
>>
>> connecting Opensips with Asterisk I can see that if a client registered on
>> Opensips server tries to make a call to a client in Asterisk domain, after
>> the INVITE, it receives a "forbidden" message from asterisk. I have set the
>> forwarding functionality in Opensips (rewriteuri function) and I'm pretty
>> sure it's something related to asterisk.
>>
>> Perhaps this is not the right section, but anyway could you help me? Do you
>> know what I should set in the sip.conf of Asterisk config file?
>>
>> Thanks a lot,
>> Carmelo
>> --
>> View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/opensips-and-asterisk-tp4962200p5003181.html
>> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>>
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>>      
> ---
> * Olle E Johansson - oej at edvina.net
> * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden
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