[Asterisk-Dev] Re: [Asterisk-cvs] asterisk/channels chan_sip.c,1.248,1.249

[Olle E. Johansson]

Thorsten Lockert wrote:

> Olle E. Johansson wrote:
> 
>>Thorsten Lockert wrote:
>>
>>>Yes, this is correct. 
>>
>>Wonderful. Thank you!
>>
>>This makes the confusion about SIP addressing disappear. It will be much
>>easier to see the difference. Great stuff.
> 
> 
> It should be noted that the old syntax of "SIP/extension at peer" is still
> valid.  And you can also now say "SIP/example.com/user", and it will be
> equivalent to "SIP/user at example.com"...  So we did not *change* how SIP
> addressing is done, but *added* an alternate syntax.

Thank you, I understood that you did not break the existing files. I like
that.

I just think that separating the two in documentation and in configuration
makes sense, and to me it certainly makes it easier to understand.

If a SIP URL looks like: sip:oej at edvina.net and I'm used to that,
the SIP/extension at peer construct confuses me. It took some time before
I understood that this was not URL dialing. Even though both ways work for
both configurations, I would suggest that using the "@" construct for
URLs and the "/" for peer with sip.conf sections is more educational.

Just brainstorming, but wouldn't a
    urldial(URL)
function make sense. This would help ENUM, since enum may return
any url - be it SIP, SIPS, H.323 or TEL.

/Olle