[asterisk-dev] [Code Review] PineTree:: Matching SIP peers beyond SIP proxy

[Klaus Darilion]

Hi Olle!

What's the status of this patch?

thanks
Klaus


12 jul 2010 kl. 16.18 skrev David Vossel:

>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/348/#review2362
> -----------------------------------------------------------
>
>
> With 1481 changes to chan_sip this is very difficult to review.   I'd understand if those were all functional changes, but 99% of what I saw looked like it was reverting things to an older version of trunk.  I don't think we want to revert some of the changes we made involving the splitting up of chan_sip.c into separate files.  I believe this patch just needs to be updated to a current version of trunk.  Otherwise it is very difficult to find your changes in all that noise.
Hmm. Something strange must have happened while producing a diff, 
because that's not really my intention... ;-)

Will try again.

/O


>
> - David
>
>
> On 2010-07-09 08:16:07, Olle E Johansson wrote:
>>
>> -----------------------------------------------------------
>> This is an automatically generated e-mail. To reply, visit:
>> https://reviewboard.asterisk.org/r/348/
>> -----------------------------------------------------------
>>
>> (Updated 2010-07-09 08:16:07)
>>
>>
>> Review request for Asterisk Developers.
>>
>>
>> Summary
>> -------
>>
>> Many of us implement asterisk behind SIP proxys for load balancing or
>> failover or both. That means that all messages to Asterisk is sent by
>> the proxy and all peer matching on IP/port fails. Asterisk simply
>> doesn't know how to separate the devices behind the proxy.
>>
>> With my new code, you can add a rule to the SIP proxy [peer] section,
>> saying "don't match me, match who sent to me". The way Asterisk does
>> that, is by reading the second VIA header. This is the device that
>> sent the message to Asterisk - another proxy or an endpoint. You can
>> also be very strict and say "match last via" - which always will be
>> the other endpoint.
>>
>> The benefit of all this is that all Asterisk features now work -
>> accountcode, codec settings, authentication. You can provision
>> different SIP trunks with different features, even though Asterisk is
>> hidden by a proxy.
>>
>> For outbound calls, you use the outbound proxy setting as before.
>>
>>
>> Diffs
>> -----
>>
>>  /trunk/channels/chan_sip.c 274865
>>
>> Diff: https://reviewboard.asterisk.org/r/348/diff
>>
>>
>> Testing
>> -------
>>
>> Testing in private networks. Have had this code in production with customer for a couple of months, albeit on 1.4.
>>
>>
>> Thanks,
>>
>> Olle E
>>
>>

---
* Olle E Johansson - oej at edvina.net
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden




-- 
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
    http://lists.digium.com/mailman/listinfo/asterisk-dev