[asterisk-users] Realtime & sip.conf

[Nicholas Blasgen]

I don't understand the
USERS vs PEER vs FRIENDS.  I just use Peer for everything.  Has to do
with "can I only contact you or can you contact me too?" ... Peer does
it all.

RealTime does have an issue.  If you don't turn on caching, then it holds no
state information.  So if you think you're going to encouter firewall issues
and need NAT=yes, then realtime will run in a static mode where you'll need
to reload each time you change anything (like a password).  I think the
proper command is something like "SIP PRUNE".

Finally, putting something like sip.conf into realtime wasn't a move I
wanted to make.  I simply generate a SIP.conf file myself via my own program
and run a SIP RELOAD (or simply reboot) each time I make a big change.
 Changes don't happen often so no biggie, where as I did want to make live
changes to other SIP users without reloading (like a person using our web
interface to change their own password).

On 12/29/07, hugolivude <hugolivude at gmail.com> wrote:
>
> Hi -
>
> I'm looking into realtime and I'm having a bit of a problem with the SIP
> part.
>
> My review of the posts seems to indicate that I should use realtime
> static for the [general] part of my sip.conf including the
> registration commands:
>
>    register=><did>:<secret>@<domain>/<did context>
>
> and use realtime realtime (funny name!) for peers and friends:
>
> [myprovider]
> type=peer
> auth=md5
> username=...
> fromuser=...
> fromdomain=...
> secret=...
> host=...
> port=5060
> nat=yes
> canreinvite=yes
> qualify=no
> disallow=all
> allow=ulaw
> dtmfmode=rfc2833
> insecure=port,invite
> context=incoming-sip
>
> Is this correct?  What's throwing me off is this statment found @
> http://www.voip-info.org/wiki/index.php?page=Asterisk%20RealTime%20Static:
>
>    NOTE: You can only store a static config OR a RealTime config. You
> cannot, for example, store
>               sip.conf and use sipfriends via RealTime.
>
> If I am correct, it would suggest that I'll have to do a reload when I
> add a DiD, but a reload won't be necessary if a new SIP client is
> added.  Do I have it right?
>
> Also, what's the difference between a peer and a user?  I used to
> think that a "user" was an agent  authorized to call in to my * box, a
> "peer" was an agent I could reach and a "freind" was both.  What's
> throwing me off now is the statement found @
>
> http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer&view_comment_id=14966
> :
>
>     With newer versions of Asterisk the concept of SIP 'users' will be
> phased out.
>
> I can't understand this especially in the context of extconfig.conf
> that uses both a sipuser and sippeer entry.  Could someone clarify for
> me?
>
> Thanks,
> H
>
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-- 
/Nick
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