[asterisk-dev] [Code Review] SIP: peer matching by callbackextension

[Alexander Harrowell]

On Friday 09 July 2010 09:50:23 Olle E. Johansson wrote:
> Vossel,
> 
> I've said that we should move the callbackexten code to a new type,
>  type=service. This will be much easier to explain for users. I also have
>  pointed to code where we register a random string and use that string to
>  match the registration on the incoming call.  That way, we'll always match
>  and send the call to the proper extension in the dialplan.
> 
> I'm doing training this week and the peers/users stuff keep coming up every
>  day, advanced engineers just have a very hard time understanding this -
>  and that's no good for our product. If you did training for a while, you
>  would also scream for these kind of issues. 

In trying to learn Asterisk, I've always found this quite odd. Obviously it's 
nice-to-have the ability to route outbound calls separately from inbound 
calls, but I would bet money on the 90% use case being "I want to have 
endpoint devices SIP-register with my Asterisk, and my Asterisk communicate 
with a service provider (whether SIP, IAX2, or trad)". So everything over time 
will end up being a "friend" as that's the convenience type.
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/pgp-signature
Size: 198 bytes
Desc: This is a digitally signed message part.
Url : http://lists.digium.com/pipermail/asterisk-dev/attachments/20100709/3f9b1861/attachment.pgp