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.<br> <br>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".
<br><br>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).
<br><br><div><span class="gmail_quote">On 12/29/07, <b class="gmail_sendername">hugolivude</b> <<a href="mailto:hugolivude@gmail.com">hugolivude@gmail.com</a>> wrote:</span><blockquote class="gmail_quote" style="margin-top: 0; margin-right: 0; margin-bottom: 0; margin-left: 0; margin-left: 0.80ex; border-left-color: #cccccc; border-left-width: 1px; border-left-style: solid; padding-left: 1ex">
Hi -<br><br>I'm looking into realtime and I'm having a bit of a problem with the SIP part.<br><br>My review of the posts seems to indicate that I should use realtime<br>static for the [general] part of my sip.conf
including the<br>registration commands:<br><br> register=><did>:<secret>@<domain>/<did context><br><br>and use realtime realtime (funny name!) for peers and friends:<br><br>[myprovider]<br>type=peer
<br>auth=md5<br>username=...<br>fromuser=...<br>fromdomain=...<br>secret=...<br>host=...<br>port=5060<br>nat=yes<br>canreinvite=yes<br>qualify=no<br>disallow=all<br>allow=ulaw<br>dtmfmode=rfc2833<br>insecure=port,invite<br>
context=incoming-sip<br><br>Is this correct? What's throwing me off is this statment found @<br><a href="http://www.voip-info.org/wiki/index.php?page=Asterisk%20RealTime%20Static">http://www.voip-info.org/wiki/index.php?page=Asterisk%20RealTime%20Static
</a>:<br><br> NOTE: You can only store a static config OR a RealTime config. You<br>cannot, for example, store<br> sip.conf and use sipfriends via RealTime.<br><br>If I am correct, it would suggest that I'll have to do a reload when I
<br>add a DiD, but a reload won't be necessary if a new SIP client is<br>added. Do I have it right?<br><br>Also, what's the difference between a peer and a user? I used to<br>think that a "user" was an agent authorized to call in to my * box, a
<br>"peer" was an agent I could reach and a "freind" was both. What's<br>throwing me off now is the statement found @<br><a href="http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer&view_comment_id=14966">
http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer&view_comment_id=14966</a>:<br><br> With newer versions of Asterisk the concept of SIP 'users' will be<br>phased out.<br><br>I can't understand this especially in the context of
extconfig.conf<br>that uses both a sipuser and sippeer entry. Could someone clarify for<br>me?<br><br>Thanks,<br>H<br><br>_______________________________________________<br>--Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--">
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</a><br></blockquote></div><br><br clear="all"><br>-- <br>/Nick