Re: [asterisk-users] Realtime & sip.conf

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@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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