[Asterisk-Dev] Asterisk and SIP phones

[Kevin P. Fleming]

Benjamin on Asterisk Mailing Lists wrote:

> That's only possible with a proxy server. Asterisk is not a proxy
> server. It acts as a user agent. So unless you implement a SIP proxy
> server module for Asterisk, you wouldn't be able to do what you want.
> I reckon that this would be quite a bit of work though.

Yes, I am aware of that :-) I don't really think I have time to 
implement chan_sip_proxy, though! There are other benefits to it, 
though, like allowing the two endpoints to negotiate codecs 
independently of Asterisk, and even renegotiate during the call if 
warranted.

 From my limited reading of the SIP RFCs and the chan_sip code, I 
believe this could be done in a limited fashion without having to build 
an entire proxy module, but I could be mistaken.

> However, the reinvite=local facility is something that would make a
> lot of sense on LANs and which can be implemented with relatively
> modest effort, since all you have to do is check if both phone's
> addresses belong to the same subnet and then allow or disallow
> reinvites acordingly. That's one particular test followed by one
> particular action, pretty straightforward.

Yes, you are correct, this would be very useful for cases where Asterisk 
does not need to be in the control path. In our case, since we are 
offering our customers very detailed CDR, they are going to want even 
"internal" calls tracked and reported, so for now we have to keep 
Asterisk in the media path as well.